From codecomplete at free.fr Sun Mar 1 10:20:54 2009 From: codecomplete at free.fr (Fred) Date: Sun, 01 Mar 2009 19:20:54 +0100 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? Message-ID: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> Hello As an easy way to show a Freeswitch server to prospects, I'm thinking of buying an Asus notebook along with a Sangom USB FXO gateway. www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port If someone's been using those two thingies, I'm curious to know if they happily run Freeswitch, or if I should look for some other hardware? Thank you. From rex.alex345 at yahoo.com Sun Mar 1 11:13:34 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST) Subject: [Freeswitch-users] To do telephony functions from web page Message-ID: <1235934814358-2405620.post@n2.nabble.com> Hi All, I am trying to do the telephonic functions(like dial, hangup, conference and etc.) from a webpage (for a customization) rather than doing it from a soft phone. What would be the optimal way of doing it? Please suggest. Thanks, Rex. -- View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405620.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/6a856e00/attachment.html From krice at freeswitch.org Sun Mar 1 11:17:39 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 01 Mar 2009 13:17:39 -0600 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <1235934814358-2405620.post@n2.nabble.com> Message-ID: Check out ESL for PHP, Perl etc, or you can use mod_xml_rpc to control things.... Both methods work well K From: Rex_Alex Reply-To: Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST) To: Subject: [Freeswitch-users] To do telephony functions from web page Hi All, I am trying to do the telephonic functions(like dial, hangup, conference and etc.) from a webpage (for a customization) rather than doing it from a soft phone. What would be the optimal way of doing it? Please suggest. Thanks, Rex. View this message in context: To do telephony functions from web page Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/da1f2954/attachment-0001.html From rex.alex345 at yahoo.com Sun Mar 1 11:51:18 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Sun, 1 Mar 2009 11:51:18 -0800 (PST) Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: References: <1235934814358-2405620.post@n2.nabble.com> Message-ID: <1235937078290-2405782.post@n2.nabble.com> Hi, Learned how to enable mod_xml_rpc but didn't find any samples. Please post me a sample to send requests(like dial) and receive responses(like uuid) from FreeSWITCH using mod_xml_rpc Please assist. Thanks, Rex. Ken Rice-2 wrote: > > Check out ESL for PHP, Perl etc, or you can use mod_xml_rpc to control > things.... Both methods work well > > K > > > > From: Rex_Alex > Reply-To: > Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST) > To: > Subject: [Freeswitch-users] To do telephony functions from web page > > Hi All, I am trying to do the telephonic functions(like dial, hangup, > conference and etc.) from a webpage (for a customization) rather than > doing > it from a soft phone. What would be the optimal way of doing it? Please > suggest. Thanks, Rex. > > View this message in context: To do telephony functions from web page > > Sent from the freeswitch-users mailing list archive > at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405782.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/eeeb7042/attachment.html From sicfslist at gmail.com Sun Mar 1 11:57:03 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 1 Mar 2009 13:57:03 -0600 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <1235937078290-2405782.post@n2.nabble.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> Message-ID: <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> Rex: The basis for xml_rpc or mod_event is something along the lines of: api $command As an example to originate a call (using xml_rpc or mod_event) you would do: api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml $context What language are you trying to do this in? SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/8edf404e/attachment.html From rex.alex345 at yahoo.com Sun Mar 1 12:10:04 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Sun, 1 Mar 2009 12:10:04 -0800 (PST) Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> Message-ID: <1235938204874-2405845.post@n2.nabble.com> Hi Shelby Ramsey, I would like to do the same in php script. Please post me a sample. Thanks, Rex. Shelby Ramsey wrote: > > Rex: > > The basis for xml_rpc or mod_event is something along the lines of: > > api $command > > As an example to originate a call (using xml_rpc or mod_event) you would > do: > > api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml $context > > What language are you trying to do this in? > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405845.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sicfslist at gmail.com Sun Mar 1 12:19:16 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 1 Mar 2009 14:19:16 -0600 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <1235938204874-2405845.post@n2.nabble.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> <1235938204874-2405845.post@n2.nabble.com> Message-ID: <35b355e90903011219p20ee2e5k9e6d553598393394@mail.gmail.com> Rex, I've never actually used PHP for this type of thing ... but you might want to start by looking here: http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/php/single_command.php?r=12216 or http://wiki.freeswitch.org/wiki/PHP_Event_Socket Good luck. I'm sure some other folks here who use PHP for this type of app will be able to assist more. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/3f2f669d/attachment.html From gmaruzz at celliax.org Sun Mar 1 12:30:25 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 1 Mar 2009 21:30:25 +0100 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? In-Reply-To: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> References: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> Message-ID: <7b197bef0903011230k16afc8d1hd43dd224570787f@mail.gmail.com> I, for one, run often FS on an eeepc900 (one year old). NEver tested max concurrent SIP calls, but for sure is able to run concurrently: - one FS - two SIP calls - two Skypiax calls - two linux Skype client instances - two Skype calls Also, I often use it to generate 6 or 8 concurrent Skype calls. So, taking account of how heavy Skype is, it probably is able to run FS with dozens concurrent SIP calls. Gm On 3/1/09, Fred wrote: > Hello > > As an easy way to show a Freeswitch server to prospects, I'm thinking > of buying an Asus notebook along with a Sangom USB FXO gateway. > > www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port > > If someone's been using those two thingies, I'm curious to know if > they happily run Freeswitch, or if I should look for some other hardware? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device 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 From csorlie at teldio.com Sun Mar 1 10:01:21 2009 From: csorlie at teldio.com (Cameron Sorlie) Date: Sun, 01 Mar 2009 13:01:21 -0500 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD Message-ID: <49AACD71.5080103@teldio.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/dc118387/attachment.html From brian at freeswitch.org Sun Mar 1 13:15:20 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Mar 2009 15:15:20 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <49AACD71.5080103@teldio.com> References: <49AACD71.5080103@teldio.com> Message-ID: <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> If i'm not mistaken those events will have a member-id in them so you can tell who they belong to. /b On Mar 1, 2009, at 12:01 PM, Cameron Sorlie wrote: > Using voice activity detection (VAD) in FreeSWITCH, how might I then > distinguish which side of a call any given TALK or NOTALK event > relates to? I am interested not just that there is activity on the > call, but interested also in which party on the call is speaking (or > not). > > Cam From Prometheus001 at gmx.net Sun Mar 1 13:38:20 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 01 Mar 2009 22:38:20 +0100 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number Message-ID: <49AB004C.6090604@gmx.net> Hello, I have the following problem while providing callback (mod_eventsocket is used): 1) I want to call a certain destination number A with a suppressed caller_id_number (this works fine with some vars in the origination string) 2) The destination number A picks up the phone and enters a target number B by DTMF 3) freeswitch then forwards the call to target number B by DTMF and I want to show the number A. I do this with uuid_setvar. The problem is that it still shows unknown. This is all with SIP. uuid_setvar however worked if I did not set the caller_id_number to unknown. Per default this is then "00000000000" and can then be changed with uuid_setvar to the number of A. But if I set caller_id_number to unknown I can no longer change it to A. Any hint? Best regards Peter From Prometheus001 at gmx.net Sun Mar 1 14:27:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 01 Mar 2009 23:27:25 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: References: <49A92BAE.4090907@gmx.net> Message-ID: <49AB0BCD.8030108@gmx.net> Hello Brian, thanks for the info. I am a step further, but it cannot load the grammar files. I am sending through event_socket: SendMsg call-command: execute execute-app-name: detect_speech execute-app-arg: pocketsphinx yes no However I get the message (also when I am using Pizza demo): 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/1000 at sip2.server.com Command Execute detect_speech(pocketsphinx yes no) 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 pocketsphinx_asr_load_grammar() Can't open language model /usr/local/freeswitch/grammar/model/communicator. 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 switch_ivr_detect_speech() Error loading Grammar 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 pocketsphinx_asr_close() Port Closed. However the grammar files are there: root at sip2:/usr/local/freeswitch/grammar/model/communicator# root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al total 12752 drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances Any hint? Best regards Peter Brian West schrieb: > You can accomplish this .... here is an example using ESL in perl > > http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 > > /b > > On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: > > >> Or back to the basics: Is it possible to use pocketsphinx through >> event >> socket? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mike at jerris.com Sun Mar 1 16:34:30 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 1 Mar 2009 19:34:30 -0500 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <1235938204874-2405845.post@n2.nabble.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> <1235938204874-2405845.post@n2.nabble.com> Message-ID: <2A727DCA-B690-4A30-97EF-4D16223ECE45@jerris.com> There are examples on the wiki for this. Mike On Mar 1, 2009, at 3:10 PM, Rex_Alex wrote: > > Hi Shelby Ramsey, > > I would like to do the same in php script. > > Please post me a sample. > > Thanks, > Rex. > > > Shelby Ramsey wrote: >> >> Rex: >> >> The basis for xml_rpc or mod_event is something along the lines of: >> >> api $command >> >> As an example to originate a call (using xml_rpc or mod_event) you >> would >> do: >> >> api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml >> $context >> >> What language are you trying to do this in? >> >> SDR >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405845.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From red.rain.seven at gmail.com Sun Mar 1 18:20:32 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Sun, 1 Mar 2009 18:20:32 -0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> Message-ID: <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> Does the freeswitch VAD is able to distinguish ring tone from human voice? The scenario is to originate a call to a IVR system(don't connect the other leg here yet) and dial DTMF to get to the designated extension number , once someone picks up and say hello ( detected by VAD) now release to connect the other leg of the call. The point is to hold the first leg till a real person picks up. If it can't be done by VAD, how should I approach this function that I want to achieve. Thanks On Sun, Mar 1, 2009 at 1:15 PM, Brian West wrote: > If i'm not mistaken those events will have a member-id in them so you > can tell who they belong to. > > /b > > On Mar 1, 2009, at 12:01 PM, Cameron Sorlie wrote: > > > Using voice activity detection (VAD) in FreeSWITCH, how might I then > > distinguish which side of a call any given TALK or NOTALK event > > relates to? I am interested not just that there is activity on the > > call, but interested also in which party on the call is speaking (or > > not). > > > > Cam > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/d4c7a68c/attachment.html From brian at freeswitch.org Sun Mar 1 18:28:18 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Mar 2009 20:28:18 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> Message-ID: <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> NO. You want something that people THINK exists and works well... Reliable human/voice detection doesn't exist in ANY form. /b On Mar 1, 2009, at 8:20 PM, Henry Huang wrote: > Does the freeswitch VAD is able to distinguish ring tone from human > voice? > The scenario is to originate a call to a IVR system(don't connect > the other leg here yet) and dial DTMF to get to the designated > extension number , once someone picks up and say hello ( detected by > VAD) now release to connect the other leg of the call. The point is > to hold the first leg till a real person picks up. > > If it can't be done by VAD, how should I approach this function that > I want to achieve. > > Thanks From red.rain.seven at gmail.com Sun Mar 1 19:05:19 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Sun, 1 Mar 2009 19:05:19 -0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com> Well, I knew it would be some future fantasy for now.. If not human detection. I guess will try to use Dialplan Tools wait for silence to wait till the ring tone is finished ,then connect the other leg. On Sun, Mar 1, 2009 at 6:28 PM, Brian West wrote: > NO. You want something that people THINK exists and works well... > Reliable human/voice detection doesn't exist in ANY form. > > /b > > On Mar 1, 2009, at 8:20 PM, Henry Huang wrote: > > > Does the freeswitch VAD is able to distinguish ring tone from human > > voice? > > The scenario is to originate a call to a IVR system(don't connect > > the other leg here yet) and dial DTMF to get to the designated > > extension number , once someone picks up and say hello ( detected by > > VAD) now release to connect the other leg of the call. The point is > > to hold the first leg till a real person picks up. > > > > If it can't be done by VAD, how should I approach this function that > > I want to achieve. > > > > Thanks > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/4d9a7260/attachment.html From brian at freeswitch.org Sun Mar 1 19:11:32 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Mar 2009 21:11:32 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com> Message-ID: Usually ringing is done in early media... so the best bet would be to ignore_early_media=true /b On Mar 1, 2009, at 9:05 PM, Henry Huang wrote: > Well, I knew it would be some future fantasy for now.. > If not human detection. I guess will try to use Dialplan Tools wait > for silence to wait till the ring tone is finished ,then connect the > other leg. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/9f2529d5/attachment.html From mrene_lists at avgs.ca Sun Mar 1 21:00:44 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 2 Mar 2009 00:00:44 -0500 Subject: [Freeswitch-users] Qt portaudio interface Message-ID: <3A65F8B6-96B4-45EE-80E9-A30AFEED0BB3@avgs.ca> Hi all, Anyone interested in contributing to a Qt interface in order to make a decent softphone using FS please reply to this thread. (also give your availability so we can have a conference call to decide stuff) Thanks, Math From kawarod at laposte.net Sun Mar 1 23:06:10 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Mar 2009 11:06:10 +0400 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test Message-ID: <49AB8562.4050806@laposte.net> Hi All, I ran some longer tests with FS 1.0.3 acting as an SBC. The test machine has the following specs: - Intel Quad Core Q9550 - 8GB RAM (far too much from what I saw) After 3 days running SIPP with 750 simultaneous calls (1500 channels) at 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. In the CLI, using status command I got this: freeswitch at internal> status UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds, 607 microseconds 15817560 session(s) since startup 22 session(s) 0/500 But when I use "show channels" or "show calls", I see nothing. So I'm wondering where are these 22 sessions ? FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs. Successful call --> 5271434 Failed call ---> 1554 (less than 0.03%) regards, rod. complete SIPP summary: ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 10.10.10.254:5060(UDP) 0 new calls during 0.856 s period 7 ms scheduler resolution 0 calls (limit 750) Peak was 750 calls, after 15 s 0 Running, 34 Paused, 0 Woken up 15544 out-of-call msg (discarded) 1 open sockets 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate (kB/s) 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate (kB/s) Messages Retrans Timeout Unexpected-Msg INVITE ----------> 5273022 0 0 100 <---------- 5273022 0 1554 180 <---------- 0 0 0 183 <---------- 0 0 0 200 <---------- E-RTD1 5271434 0 0 ACK ----------> 5271434 0 Pause [ 35.0s] 5271434 0 BYE ----------> 5271434 0 0 200 <---------- 5271434 0 0 ------------------------------ Test Terminated -------------------------------- ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2009-02-27 09:11:31 Last Reset Time | 2009-03-02 07:49:10 Current Time | 2009-03-02 07:49:11 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:857 | 70:37:39:429 Call Rate | 0.000 cps | 20.739 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 5273022 Total Call created | | 5273022 Current Call | 34 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 5271434 Failed call | 0 | 1554 -------------------------+---------------------------+-------------------------- Response Time 1 | 00:00:00:000 | 00:00:00:240 Call Length | 38:32:13:386 | 00:00:36:131 ------------------------------ Test Terminated -------------------------------- From red.rain.seven at gmail.com Sun Mar 1 23:18:02 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Sun, 1 Mar 2009 23:18:02 -0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com> Message-ID: <59ad9ca10903012318k53852016ied8d982a467577c3@mail.gmail.com> ignore_early_media=true is not going to do the trick since once the IVR picks up the call on leg A, the ring tone is stopped and the IVR is going to play pre-recorded voice menu. And the freeswtich is going to send DTMF to reach a certain extension number say 101. Then the ring tone is going to start again while the IVR is going to dial the 101 extension(or even play moh while dialing). After extension 101 picks up, this is when I want the "originate" to connect call leg B on some other number. On Sun, Mar 1, 2009 at 7:11 PM, Brian West wrote: > Usually ringing is done in early media... so the best bet would be to > ignore_early_media=true > /b > > On Mar 1, 2009, at 9:05 PM, Henry Huang wrote: > > Well, I knew it would be some future fantasy for now.. > If not human detection. I guess will try to use Dialplan Tools wait for > silence to wait till the ring tone is finished ,then connect the other leg. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/ebab2445/attachment-0001.html From saigop at gmail.com Mon Mar 2 01:25:23 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Mon, 2 Mar 2009 14:55:23 +0530 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <2A727DCA-B690-4A30-97EF-4D16223ECE45@jerris.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> <1235938204874-2405845.post@n2.nabble.com> <2A727DCA-B690-4A30-97EF-4D16223ECE45@jerris.com> Message-ID: <2ea4d47e0903020125m4be4a5ffl4c33e6f3325a2919@mail.gmail.com> Hi Rex, Please find the attached file for the PHP script. This script has been executed in FS 1.0.2. put these two scripts in htdocs directory. access the http://localhost/sample2.php so that two text box will appear. you can able to give the extension number and mobile number to dial. Try this :) On Mon, Mar 2, 2009 at 6:04 AM, Michael Jerris wrote: > There are examples on the wiki for this. > > Mike > > On Mar 1, 2009, at 3:10 PM, Rex_Alex wrote: > > > > > Hi Shelby Ramsey, > > > > I would like to do the same in php script. > > > > Please post me a sample. > > > > Thanks, > > Rex. > > > > > > Shelby Ramsey wrote: > >> > >> Rex: > >> > >> The basis for xml_rpc or mod_event is something along the lines of: > >> > >> api $command > >> > >> As an example to originate a call (using xml_rpc or mod_event) you > >> would > >> do: > >> > >> api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml > >> $context > >> > >> What language are you trying to do this in? > >> > >> SDR > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > -- > > View this message in context: > http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405845.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/ae5592d0/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sample2.php Type: application/octet-stream Size: 405 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/ae5592d0/attachment.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: testsample.php Type: application/octet-stream Size: 1434 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/ae5592d0/attachment-0001.obj From gopal2krishnan at gmail.com Mon Mar 2 01:29:13 2009 From: gopal2krishnan at gmail.com (Gopal krishnan) Date: Mon, 2 Mar 2009 14:59:13 +0530 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <1235740392995-2395557.post@n2.nabble.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <49A6E1DB.3070806@freeswitch.org> <1235740392995-2395557.post@n2.nabble.com> Message-ID: <2ea4d47e0903020129y676bbc35n8727f8906237edce@mail.gmail.com> Hi, Actually what is the difference between ESL in FS 1.0.3 and event socket in FS 1.0.2. Is the FS 1.0.3 ESL superior? On Fri, Feb 27, 2009 at 6:43 PM, Rex_Alex wrote: > Hi All, I did what you have all suggested. Now its working perfectly. > Thanks a lot for all your assistance. Rex. > > Raymond Chandler wrote: > and it will probably be a good idea to do make phpmod-install so that the > .so and .php files gets into the correct place to be included -Ray Mathieu > Rene wrote: > > You need your distro's php dev pakage. > On 26-Feb-09, at > 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh && > ./configure && make >> install and did Mathieu's suggestion but getting > error as below, >> [root at server esl]# make phpmod make MYLIB="../libesl.a" > >> SOLINK="-shared -Xlinker -x" >> > CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >> -g > -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >> -Werror > -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> > -Wmissing-prototypes" >> > CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >> -DHAVE_EDITLINE > -g -ggdb -I../../libs/libedit/src/ -fPIC" >> CXX_CFLAGS="" -C php make[1]: > php-config: Command not found make[1]: >> Entering directory > `/root/freeswitch-1.0.3/libs/esl/php' g++ >> > -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >> -ggdb > -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >> > esl_wrap.cpp:717:18: error: zend.h: No such file or directory >> > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >> > esl_wrap.cpp:719:17: error: php.h: No such file or directory >> > esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >> > esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >> > directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >> > scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >> > ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >> > ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >> error: > expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >> variable or > field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error: > ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was > not declared in this scope >> esl_wrap.cpp:793: error: expected > primary-expression before ?void? >> esl_wrap.cpp:793: error: expected > primary-expression before ?*? token >> esl_wrap.cpp:793: error: ?type? was > not declared in this scope >> esl_wrap.cpp:793: error: expected > primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer > expression list treated as >> compound expression esl_wrap.cpp:793: error: > expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 > make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: > *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i > wrong? Thanks, Rex >> >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34 > AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it.. > Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh && > ./configure && >> make install And then do Mathieu's suggestion? -MC >> > _______________________________________________ Freeswitch-users >> mailing > list Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>> > http://www.freeswitch.org >> >> >> > ------------------------------------------------------------------------ >> > View this message in context: Re: ESL Wrapper >> >> Sent from the > freeswitch-users mailing list archive >> at Nabble.com. >> > _______________________________________________ >> Freeswitch-users mailing > list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > View this message in context: Re: ESL Wrapper > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/fb354cfa/attachment.html From gopal2krishnan at gmail.com Mon Mar 2 01:42:19 2009 From: gopal2krishnan at gmail.com (Gopal krishnan) Date: Mon, 2 Mar 2009 15:12:19 +0530 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? In-Reply-To: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> References: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> Message-ID: <2ea4d47e0903020142n62479f72o996243ea6b5bee64@mail.gmail.com> Hi Fred, Yes you can use Sangoma USB FXO with your laptop. You need to install openzap for this. But for testing you can use this driver. Still there is some issue with Openzap with FS as for as I used. while installing Sangoma USB FXO device you need to use beta drivers. On Sun, Mar 1, 2009 at 11:50 PM, Fred wrote: > Hello > > As an easy way to show a Freeswitch server to prospects, I'm thinking > of buying an Asus notebook along with a Sangom USB FXO gateway. > > www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port > > If someone's been using those two thingies, I'm curious to know if > they happily run Freeswitch, or if I should look for some other hardware? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/d36552a8/attachment-0001.html From codecomplete at free.fr Mon Mar 2 03:47:19 2009 From: codecomplete at free.fr (Fred) Date: Mon, 02 Mar 2009 12:47:19 +0100 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? In-Reply-To: References: Message-ID: <7.0.1.0.2.20090302124612.028657a8@free.fr> Thanks guys for the feedback. So, the OpenZap driver isn't ready for production yet? From sridhart at alcatel-lucent.com Mon Mar 2 03:52:10 2009 From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar)) Date: Mon, 2 Mar 2009 17:22:10 +0530 Subject: [Freeswitch-users] Running freeswitch on powerpc In-Reply-To: References: Message-ID: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> Hi all, I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code Regards Sridhar > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On > Behalf Of freeswitch-users-request at lists.freeswitch.org > Sent: Monday, February 02, 2009 9:12 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 32, Issue 17 > > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more > specific than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: Call Variable not available when call hangup (shehzad p) > 2. Re: How do I set my FS internal ip address to a "static" > value. (clif at eugeneweb.com) > 3. Re: Call Variable not available when call hangup > (Anthony Minessale) > 4. Re: How do I set my FS internal ip address to a "static" > value. (Brian West) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST) > From: shehzad p > Subject: Re: [Freeswitch-users] Call Variable not available when call > hangup > To: freeswitch-users at lists.freeswitch.org > Message-ID: <21791503.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > > one question is that when javascript is being called from > dial plan, I get the session object already available, It is > for A leg of channel, So when javascript is called after > Bridge how can I get the session object for B leg also? > > > Anthony Minessale-2 wrote: > > > > the leg you are running the script on is not hungup, the > other leg of the > > call is. > > > > If it was hungup you would not be executing the script. > > > > Asterisk and the h ext and the whole dead-agi thing are all > poor design > > showing it's teeth. > > We do not support anything like it. > > > > > > You can however try this: (see the link below) > > > > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks > -giving-me-headaches-p21614840.html > > > > > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: > > > >> > >> Is there any settings that when call hangup control can be > transferred to > >> another context and these CDR values can be accessible > there? (just like > >> in > >> Asterisk, h extension) > >> > >> shehzad p wrote: > >> > > >> > Hi all, > >> > > >> > I need to process some CDR variables in Dialplan, like > call duration, > >> > Answered time etc. > >> > but when I place info application after bridge, it is > not listing them > >> > properly as below: > >> > =========================================== > >> > Caller-Channel-Created-Time: [1233573341672157] > >> > Caller-Channel-Answered-Time: [1233573342712939] > >> > Caller-Channel-Hangup-Time: [0] > >> > ========================================== > >> > Here Hangup time is 0, So how can I find actual values? > >> > > >> > --I know that we can use xml_cdr or cdr_csv, but my > current need is to > >> get > >> > those values from dialplan itself so that can be passed to some > >> script... > >> > > >> > > >> > thanks, > >> > msp > >> > > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21789152.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com ny.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 f%2B888 at conference.freeswitch.org> > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21791503.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 2 > Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST) > From: clif at eugeneweb.com > Subject: Re: [Freeswitch-users] How do I set my FS internal ip address > to a "static" value. > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > Hi Gang, > > I've been struggleing with this also. Actually I can get it > to bind to my > address, the problem is it randomly drops my calls. :-( > > I have a FS running on a box with a static IP and I can start > a call between > two extensions and it will go for hours. Then I add anther > interface say eth0:0 > with a new static IP and reconfigure my phones and FS to use > that, and the > calls drop after about 15-20 mins. Though it's pretty random. > > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and > freeswitch-1.0.1. > Here is my /etc/network/interfaces: > > # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) > > # The loopback interface > auto lo > iface lo inet loopback > > # The first network card - this entry was created during the Debian > installation > auto eth0 eth0:0 > iface eth0 inet dhcp > iface eth0:0 inet static > address 192.168.0.249 > netmask 255.255.255.0 > gateway 192.168.0.254 > > The only change I made to the FS config is in Vars.xml. I > added this line close > to the top: > > > > Here is the console log of the call being dropped: > > freeswitch at archive> sofia status > API CALL [sofia(status)] output: > Name Type > Data > State > ============================================================== > =================================== > external profile > sip:mod_sofia at 67.171.158.226:5080 > RUNNING (0) > internal profile > sip:mod_sofia at 192.168.0.249:5060 > RUNNING (2) > nat profile > sip:mod_sofia at 67.171.158.226:5070 > RUNNING (0) > default alias > internal > ALIASED > outbound alias > external > ALIASED > 192.168.0.249 alias > internal > ALIASED > ============================================================== > =================================== > 3 profiles 3 aliases > > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 > sofia_glue_restart_all_profiles() Reload XML [Success] > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() > ENUM Reloaded > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 > sofia_read_frame() Hangup > sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp > ;fs_nat=yes > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 > switch_ivr_multi_threaded_bridge() Hangup > sofia/internal/1001 at 192.168.0.249 > [CS_EXECUTE] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 6 > (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud > p;fs_nat=yes) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel > sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp > ;fs_nat=yes > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 5 > (sofia/internal/1001 at 192.168.0.249) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel > sofia/internal/1001 at 192.168.0.249 > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [192.168.0.249] for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() > Adding Alias [default] > for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() > Started Profile > internal [sofia_reg_internal] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [outbound] for profile [external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() > Started Profile > external [sofia_reg_external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:645 > sofia_profile_thread_run() waiting for > worker thread > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() > Started Profile nat > [sofia_reg_nat] > sofia status > API CALL [sofia(status)] output: > Name Type > Data > State > ============================================================== > =================================== > external profile > sip:mod_sofia at 67.171.158.226:5080 > RUNNING (0) > internal profile > sip:mod_sofia at 192.168.0.249:5060 > RUNNING (0) > outbound alias > external > ALIASED > 192.168.0.249 alias > internal > ALIASED > nat profile > sip:mod_sofia at 67.171.158.226:5070 > RUNNING (0) > default alias > internal > ALIASED > ============================================================== > =================================== > 3 profiles 3 aliases > > There is an older thread that says one should set > > but in this (later) thread is says only Jingleling usese that > variable. > ie. see: > http://www.mail-archive.com/freeswitch-users at lists.freeswitch. > org/msg00695.html > http://www.mail-archive.com/freeswitch-users at lists.freeswitch. > org/msg07345.html > > So what do you think causes this? What is the correct way? ;-) > > > Thanks, > Clif > > > > > ------------------------------ > > Message: 3 > Date: Mon, 2 Feb 2009 09:41:05 -0600 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Call Variable not available when call > hangup > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > you can't that's why i said it was a horrible approach. > That's also why i posted you the instructions on the only > elegant solution > to your problem. > > > On Mon, Feb 2, 2009 at 9:21 AM, shehzad p wrote: > > > > > > > one question is that when javascript is being called from > dial plan, I get > > the session object already available, It is for A leg of channel, > > So when javascript is called after Bridge how can I get the > session object > > for B leg also? > > > > > > Anthony Minessale-2 wrote: > > > > > > the leg you are running the script on is not hungup, the > other leg of the > > > call is. > > > > > > If it was hungup you would not be executing the script. > > > > > > Asterisk and the h ext and the whole dead-agi thing are > all poor design > > > showing it's teeth. > > > We do not support anything like it. > > > > > > > > > You can however try this: (see the link below) > > > > > > > > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks > -giving-me-headaches-p21614840.html > > > > > > > > > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p > wrote: > > > > > >> > > >> Is there any settings that when call hangup control can > be transferred > > to > > >> another context and these CDR values can be accessible > there? (just like > > >> in > > >> Asterisk, h extension) > > >> > > >> shehzad p wrote: > > >> > > > >> > Hi all, > > >> > > > >> > I need to process some CDR variables in Dialplan, like > call duration, > > >> > Answered time etc. > > >> > but when I place info application after bridge, it is > not listing them > > >> > properly as below: > > >> > =========================================== > > >> > Caller-Channel-Created-Time: [1233573341672157] > > >> > Caller-Channel-Answered-Time: [1233573342712939] > > >> > Caller-Channel-Hangup-Time: [0] > > >> > ========================================== > > >> > Here Hangup time is 0, So how can I find actual values? > > >> > > > >> > --I know that we can use xml_cdr or cdr_csv, but my > current need is to > > >> get > > >> > those values from dialplan itself so that can be passed to some > > >> script... > > >> > > > >> > > > >> > thanks, > > >> > msp > > >> > > > >> > > >> -- > > >> View this message in context: > > >> > > > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21789152.html > > >> Sent from the Freeswitch-users mailing list archive at > Nabble.com. > > >> > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > < > > > MSN%3Aanthony_minessale at hotmail.com hotmail.com> > > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com ny.minessale at gmail.com> > > > sale at gmail.com> > > > > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > < > > > sip%3A888 at conference.freeswitch.org eswitch.org> > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org f%2B888 at conference.freeswitch.org> > > > 253Aconf%252B888 at conference.freeswitch.org> > > > > > > pstn:213-799-1400 > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > > http://www.freeswitch.org > > > > > > > > > > -- > > View this message in context: > > > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21791503.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com ny.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 f%2B888 at conference.freeswitch.org> > pstn:213-799-1400 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachm > ents/20090202/2d430e44/attachment-0001.html > > ------------------------------ > > Message: 4 > Date: Mon, 2 Feb 2009 09:41:39 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] How do I set my FS internal ip address > to a "static" value. > To: freeswitch-users at lists.freeswitch.org > Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > you need to add this setting to sofia.conf.xml > > > > > You'll also need to edit the sofia profiles and input the > exact IP you > wish it to bind to. The params are sip-ip and rtp-ip. > > /b > > On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote: > > > Hi Gang, > > > > I've been struggleing with this also. Actually I can get it > to bind > > to my > > address, the problem is it randomly drops my calls. :-( > > > > I have a FS running on a box with a static IP and I can > start a call > > between > > two extensions and it will go for hours. Then I add anther > interface > > say eth0:0 > > with a new static IP and reconfigure my phones and FS to use that, > > and the > > calls drop after about 15-20 mins. Though it's pretty random. > > > > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and > > freeswitch-1.0.1. > > Here is my /etc/network/interfaces: > > > > # /etc/network/interfaces -- configuration file for > ifup(8), ifdown(8) > > > > # The loopback interface > > auto lo > > iface lo inet loopback > > > > # The first network card - this entry was created during the Debian > > installation > > auto eth0 eth0:0 > > iface eth0 inet dhcp > > iface eth0:0 inet static > > address 192.168.0.249 > > netmask 255.255.255.0 > > gateway 192.168.0.254 > > > > The only change I made to the FS config is in Vars.xml. I > added this > > line close > > to the top: > > > > > > > > Here is the console log of the call being dropped: > > > > freeswitch at archive> sofia status > > API CALL [sofia(status)] output: > > Name Type > Data > > State > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > > ====================================================================== > > external profile > sip:mod_sofia at 67.171.158.226:5080 > > RUNNING (0) > > internal profile > sip:mod_sofia at 192.168.0.249:5060 > > RUNNING (2) > > nat profile > sip:mod_sofia at 67.171.158.226:5070 > > RUNNING (0) > > default alias > internal > > ALIASED > > outbound alias > external > > ALIASED > > 192.168.0.249 alias > internal > > ALIASED > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > > ====================================================================== > > 3 profiles 3 aliases > > > > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 > > sofia_glue_restart_all_profiles() Reload XML [Success] > > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM > > Reloaded > > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 > sofia_read_frame() Hangup > > sofia/internal/ > > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 > > switch_ivr_multi_threaded_bridge() Hangup > sofia/internal/1001 at 192.168.0.249 > > [CS_EXECUTE] [NORMAL_CLEARING] > > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > > switch_core_session_thread() Session 6 > > (sofia/internal/ > > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) > > Ended > > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > > switch_core_session_thread() Close Channel > > sofia/internal/ > > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes > > [CS_HANGUP] > > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > > switch_core_session_thread() Session 5 > (sofia/internal/1001 at 192.168.0.249 > > ) > > Ended > > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > > switch_core_session_thread() Close Channel > sofia/internal/1001 at 192.168.0.249 > > [CS_HANGUP] > > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() > > waiting for > > worker thread > > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() > > waiting for > > worker thread > > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() > Adding Alias > > [192.168.0.249] for profile [internal] > > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding > > Alias [default] > > for profile [internal] > > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started > > Profile > > internal [sofia_reg_internal] > > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() > Adding Alias > > [outbound] for profile [external] > > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started > > Profile > > external [sofia_reg_external] > > 2009-02-01 13:23:20 [NOTICE] sofia.c:645 > sofia_profile_thread_run() > > waiting for > > worker thread > > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started > > Profile nat > > [sofia_reg_nat] > > sofia status > > API CALL [sofia(status)] output: > > Name Type > Data > > State > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > > ====================================================================== > > external profile > sip:mod_sofia at 67.171.158.226:5080 > > RUNNING (0) > > internal profile > sip:mod_sofia at 192.168.0.249:5060 > > RUNNING (0) > > outbound alias > external > > ALIASED > > 192.168.0.249 alias > internal > > ALIASED > > nat profile > sip:mod_sofia at 67.171.158.226:5070 > > RUNNING (0) > > default alias > internal > > ALIASED > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > > ====================================================================== > > 3 profiles 3 aliases > > > > There is an older thread that says one should set > > > > but in this (later) thread is says only Jingleling usese that > > variable. > > ie. see: > > > http://www.mail-archive.com/freeswitch-users at lists.freeswitch. > org/msg00695.html > > > http://www.mail-archive.com/freeswitch-users at lists.freeswitch. > org/msg07345.html > > > > So what do you think causes this? What is the correct way? ;-) > > > > > > Thanks, > > Clif > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > http://www.freeswitch.org > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 32, Issue 17 > ************************************************ > From gmaruzz at celliax.org Mon Mar 2 04:11:26 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 2 Mar 2009 13:11:26 +0100 Subject: [Freeswitch-users] Running freeswitch on powerpc In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <7b197bef0903020411q72be83efxace263ab3401c001@mail.gmail.com> On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar) wrote: > I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code Hi Sridhar, I don't think someone has tried that. It will probably be you that let us all know which (if any) changes needs to be done. :-) 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 Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar) wrote: > Hi all, > > I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code > > Regards > Sridhar > > >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On >> Behalf Of freeswitch-users-request at lists.freeswitch.org >> Sent: Monday, February 02, 2009 9:12 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Freeswitch-users Digest, Vol 32, Issue 17 >> >> Send Freeswitch-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more >> specific than "Re: Contents of Freeswitch-users digest..." >> >> >> Today's Topics: >> >> 1. Re: Call Variable not available when call hangup (shehzad p) >> 2. Re: How do I set my FS internal ip address to a "static" >> value. (clif at eugeneweb.com) >> 3. Re: Call Variable not available when call hangup >> (Anthony Minessale) >> 4. Re: How do I set my FS internal ip address to a "static" >> value. (Brian West) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST) >> From: shehzad p >> Subject: Re: [Freeswitch-users] Call Variable not available when call >> hangup >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <21791503.post at talk.nabble.com> >> Content-Type: text/plain; charset=us-ascii >> >> >> >> one question is that when javascript is being called from >> dial plan, I get the session object already available, It is >> for A leg of channel, So when javascript is called after >> Bridge how can I get the session object for B leg also? >> >> >> Anthony Minessale-2 wrote: >> > >> > the leg you are running the script on is not hungup, the >> other leg of the >> > call is. >> > >> > If it was hungup you would not be executing the script. >> > >> > Asterisk and the h ext and the whole dead-agi thing are all >> poor design >> > showing it's teeth. >> > We do not support anything like it. >> > >> > >> > You can however try this: (see the link below) >> > >> > >> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks >> -giving-me-headaches-p21614840.html >> > >> > >> > >> > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: >> > >> >> >> >> Is there any settings that when call hangup control can be >> transferred to >> >> another context and these CDR values can be accessible >> there? (just like >> >> in >> >> Asterisk, h extension) >> >> >> >> shehzad p wrote: >> >> > >> >> > Hi all, >> >> > >> >> > I need to process some CDR variables in Dialplan, like >> call duration, >> >> > Answered time etc. >> >> > but when I place info application after bridge, it is >> not listing them >> >> > properly as below: >> >> > =========================================== >> >> > Caller-Channel-Created-Time: [1233573341672157] >> >> > Caller-Channel-Answered-Time: [1233573342712939] >> >> > Caller-Channel-Hangup-Time: [0] >> >> > ========================================== >> >> > Here Hangup time is 0, So how can I find actual values? >> >> > >> >> > --I know that we can use xml_cdr or cdr_csv, but my >> current need is to >> >> get >> >> > those values from dialplan itself so that can be passed to some >> >> script... >> >> > >> >> > >> >> > thanks, >> >> > msp >> >> > >> >> >> >> -- >> >> View this message in context: >> >> >> http://www.nabble.com/Call-Variable-not-available-when-call-ha >> ngup-tp21788550p21789152.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com> ny.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> f%2B888 at conference.freeswitch.org> >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Call-Variable-not-available-when-call-ha >> ngup-tp21788550p21791503.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> ------------------------------ >> >> Message: 2 >> Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST) >> From: clif at eugeneweb.com >> Subject: Re: [Freeswitch-users] How do I set my FS internal ip address >> to a "static" value. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed >> >> Hi Gang, >> >> I've been struggleing with this also. Actually I can get it >> to bind to my >> address, the problem is it randomly drops my calls. :-( >> >> I have a FS running on a box with a static IP and I can start >> a call between >> two extensions and it will go for hours. Then I add anther >> interface say eth0:0 >> with a new static IP and reconfigure my phones and FS to use >> that, and the >> calls drop after about 15-20 mins. Though it's pretty random. >> >> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and >> freeswitch-1.0.1. >> Here is my /etc/network/interfaces: >> >> # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) >> >> # The loopback interface >> auto lo >> iface lo inet loopback >> >> # The first network card - this entry was created during the Debian >> installation >> auto eth0 eth0:0 >> iface eth0 inet dhcp >> iface eth0:0 inet static >> address 192.168.0.249 >> netmask 255.255.255.0 >> gateway 192.168.0.254 >> >> The only change I made to the FS config is in Vars.xml. I >> added this line close >> to the top: >> >> >> >> Here is the console log of the call being dropped: >> >> freeswitch at archive> sofia status >> API CALL [sofia(status)] output: >> Name Type >> Data >> State >> ============================================================== >> =================================== >> external profile >> sip:mod_sofia at 67.171.158.226:5080 >> RUNNING (0) >> internal profile >> sip:mod_sofia at 192.168.0.249:5060 >> RUNNING (2) >> nat profile >> sip:mod_sofia at 67.171.158.226:5070 >> RUNNING (0) >> default alias >> internal >> ALIASED >> outbound alias >> external >> ALIASED >> 192.168.0.249 alias >> internal >> ALIASED >> ============================================================== >> =================================== >> 3 profiles 3 aliases >> >> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 >> sofia_glue_restart_all_profiles() Reload XML [Success] >> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() >> ENUM Reloaded >> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 >> sofia_read_frame() Hangup >> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp >> ;fs_nat=yes >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 >> switch_ivr_multi_threaded_bridge() Hangup >> sofia/internal/1001 at 192.168.0.249 >> [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >> switch_core_session_thread() Session 6 >> (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud >> p;fs_nat=yes) >> Ended >> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >> switch_core_session_thread() Close Channel >> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp >> ;fs_nat=yes >> [CS_HANGUP] >> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >> switch_core_session_thread() Session 5 >> (sofia/internal/1001 at 192.168.0.249) >> Ended >> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >> switch_core_session_thread() Close Channel >> sofia/internal/1001 at 192.168.0.249 >> [CS_HANGUP] >> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() waiting for >> worker thread >> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() waiting for >> worker thread >> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias >> [192.168.0.249] for profile [internal] >> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() >> Adding Alias [default] >> for profile [internal] >> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() >> Started Profile >> internal [sofia_reg_internal] >> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias >> [outbound] for profile [external] >> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() >> Started Profile >> external [sofia_reg_external] >> 2009-02-01 13:23:20 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() waiting for >> worker thread >> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() >> Started Profile nat >> [sofia_reg_nat] >> sofia status >> API CALL [sofia(status)] output: >> Name Type >> Data >> State >> ============================================================== >> =================================== >> external profile >> sip:mod_sofia at 67.171.158.226:5080 >> RUNNING (0) >> internal profile >> sip:mod_sofia at 192.168.0.249:5060 >> RUNNING (0) >> outbound alias >> external >> ALIASED >> 192.168.0.249 alias >> internal >> ALIASED >> nat profile >> sip:mod_sofia at 67.171.158.226:5070 >> RUNNING (0) >> default alias >> internal >> ALIASED >> ============================================================== >> =================================== >> 3 profiles 3 aliases >> >> There is an older thread that says one should set >> >> but in this (later) thread is says only Jingleling usese that >> variable. >> ie. see: >> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >> org/msg00695.html >> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >> org/msg07345.html >> >> So what do you think causes this? What is the correct way? ;-) >> >> >> Thanks, >> Clif >> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Mon, 2 Feb 2009 09:41:05 -0600 >> From: Anthony Minessale >> Subject: Re: [Freeswitch-users] Call Variable not available when call >> hangup >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> you can't that's why i said it was a horrible approach. >> That's also why i posted you the instructions on the only >> elegant solution >> to your problem. >> >> >> On Mon, Feb 2, 2009 at 9:21 AM, shehzad p wrote: >> >> > >> > >> > one question is that when javascript is being called from >> dial plan, I get >> > the session object already available, It is for A leg of channel, >> > So when javascript is called after Bridge how can I get the >> session object >> > for B leg also? >> > >> > >> > Anthony Minessale-2 wrote: >> > > >> > > the leg you are running the script on is not hungup, the >> other leg of the >> > > call is. >> > > >> > > If it was hungup you would not be executing the script. >> > > >> > > Asterisk and the h ext and the whole dead-agi thing are >> all poor design >> > > showing it's teeth. >> > > We do not support anything like it. >> > > >> > > >> > > You can however try this: (see the link below) >> > > >> > > >> > >> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks >> -giving-me-headaches-p21614840.html >> > > >> > > >> > > >> > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p >> wrote: >> > > >> > >> >> > >> Is there any settings that when call hangup control can >> be transferred >> > to >> > >> another context and these CDR values can be accessible >> there? (just like >> > >> in >> > >> Asterisk, h extension) >> > >> >> > >> shehzad p wrote: >> > >> > >> > >> > Hi all, >> > >> > >> > >> > I need to process some CDR variables in Dialplan, like >> call duration, >> > >> > Answered time etc. >> > >> > but when I place info application after bridge, it is >> not listing them >> > >> > properly as below: >> > >> > =========================================== >> > >> > Caller-Channel-Created-Time: [1233573341672157] >> > >> > Caller-Channel-Answered-Time: [1233573342712939] >> > >> > Caller-Channel-Hangup-Time: [0] >> > >> > ========================================== >> > >> > Here Hangup time is 0, So how can I find actual values? >> > >> > >> > >> > --I know that we can use xml_cdr or cdr_csv, but my >> current need is to >> > >> get >> > >> > those values from dialplan itself so that can be passed to some >> > >> script... >> > >> > >> > >> > >> > >> > thanks, >> > >> > msp >> > >> > >> > >> >> > >> -- >> > >> View this message in context: >> > >> >> > >> http://www.nabble.com/Call-Variable-not-available-when-call-ha >> ngup-tp21788550p21789152.html >> > >> Sent from the Freeswitch-users mailing list archive at >> Nabble.com. >> > >> >> > >> >> > >> _______________________________________________ >> > >> Freeswitch-users mailing list >> > >> Freeswitch-users at lists.freeswitch.org >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE: >> > http://lists.freeswitch.org/mailman/options/freeswitch-users >> > >> http://www.freeswitch.org >> > >> >> > > >> > > >> > > >> > > -- >> > > Anthony Minessale II >> > > >> > > FreeSWITCH http://www.freeswitch.org/ >> > > ClueCon http://www.cluecon.com/ >> > > >> > > AIM: anthm >> > > MSN:anthony_minessale at hotmail.com >> < >> > >> MSN%3Aanthony_minessale at hotmail.com> hotmail.com> >> > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com> ny.minessale at gmail.com> >> > >> > sale at gmail.com> >> > > >> > > IRC: irc.freenode.net #freeswitch >> > > >> > > FreeSWITCH Developer Conference >> > > sip:888 at conference.freeswitch.org >> < >> > >> sip%3A888 at conference.freeswitch.org> eswitch.org> >> > > >> > > iax:guest at conference.freeswitch.org/888 >> > > >> googletalk:conf+888 at conference.freeswitch.org> f%2B888 at conference.freeswitch.org> >> > >> > 253Aconf%252B888 at conference.freeswitch.org> >> > > >> > > pstn:213-799-1400 >> > > >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > Freeswitch-users at lists.freeswitch.org >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> > > http://www.freeswitch.org >> > > >> > > >> > >> > -- >> > View this message in context: >> > >> http://www.nabble.com/Call-Variable-not-available-when-call-ha >> ngup-tp21788550p21791503.html >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com> ny.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> f%2B888 at conference.freeswitch.org> >> pstn:213-799-1400 >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachm >> ents/20090202/2d430e44/attachment-0001.html >> >> ------------------------------ >> >> Message: 4 >> Date: Mon, 2 Feb 2009 09:41:39 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] How do I set my FS internal ip address >> to a "static" value. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org> >> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes >> >> you need to add this setting to sofia.conf.xml >> >> >> >> >> You'll also need to edit the sofia profiles and input the >> exact IP you >> wish it to bind to. The params are sip-ip and rtp-ip. >> >> /b >> >> On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote: >> >> > Hi Gang, >> > >> > I've been struggleing with this also. Actually I can get it >> to bind >> > to my >> > address, the problem is it randomly drops my calls. :-( >> > >> > I have a FS running on a box with a static IP and I can >> start a call >> > between >> > two extensions and it will go for hours. Then I add anther >> interface >> > say eth0:0 >> > with a new static IP and reconfigure my phones and FS to use that, >> > and the >> > calls drop after about 15-20 mins. Though it's pretty random. >> > >> > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and >> > freeswitch-1.0.1. >> > Here is my /etc/network/interfaces: >> > >> > # /etc/network/interfaces -- configuration file for >> ifup(8), ifdown(8) >> > >> > # The loopback interface >> > auto lo >> > iface lo inet loopback >> > >> > # The first network card - this entry was created during the Debian >> > installation >> > auto eth0 eth0:0 >> > iface eth0 inet dhcp >> > iface eth0:0 inet static >> > address 192.168.0.249 >> > netmask 255.255.255.0 >> > gateway 192.168.0.254 >> > >> > The only change I made to the FS config is in Vars.xml. I >> added this >> > line close >> > to the top: >> > >> > >> > >> > Here is the console log of the call being dropped: >> > >> > freeswitch at archive> sofia status >> > API CALL [sofia(status)] output: >> > Name Type >> Data >> > State >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > >> ====================================================================== >> > external profile >> sip:mod_sofia at 67.171.158.226:5080 >> > RUNNING (0) >> > internal profile >> sip:mod_sofia at 192.168.0.249:5060 >> > RUNNING (2) >> > nat profile >> sip:mod_sofia at 67.171.158.226:5070 >> > RUNNING (0) >> > default alias >> internal >> > ALIASED >> > outbound alias >> external >> > ALIASED >> > 192.168.0.249 alias >> internal >> > ALIASED >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > >> ====================================================================== >> > 3 profiles 3 aliases >> > >> > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 >> > sofia_glue_restart_all_profiles() Reload XML [Success] >> > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM >> > Reloaded >> > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 >> sofia_read_frame() Hangup >> > sofia/internal/ >> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 >> > switch_ivr_multi_threaded_bridge() Hangup >> sofia/internal/1001 at 192.168.0.249 >> > [CS_EXECUTE] [NORMAL_CLEARING] >> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >> > switch_core_session_thread() Session 6 >> > (sofia/internal/ >> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) >> > Ended >> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >> > switch_core_session_thread() Close Channel >> > sofia/internal/ >> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes >> > [CS_HANGUP] >> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >> > switch_core_session_thread() Session 5 >> (sofia/internal/1001 at 192.168.0.249 >> > ) >> > Ended >> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >> > switch_core_session_thread() Close Channel >> sofia/internal/1001 at 192.168.0.249 >> > [CS_HANGUP] >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() >> > waiting for >> > worker thread >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() >> > waiting for >> > worker thread >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() >> Adding Alias >> > [192.168.0.249] for profile [internal] >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding >> > Alias [default] >> > for profile [internal] >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started >> > Profile >> > internal [sofia_reg_internal] >> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() >> Adding Alias >> > [outbound] for profile [external] >> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started >> > Profile >> > external [sofia_reg_external] >> > 2009-02-01 13:23:20 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() >> > waiting for >> > worker thread >> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started >> > Profile nat >> > [sofia_reg_nat] >> > sofia status >> > API CALL [sofia(status)] output: >> > Name Type >> Data >> > State >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > >> ====================================================================== >> > external profile >> sip:mod_sofia at 67.171.158.226:5080 >> > RUNNING (0) >> > internal profile >> sip:mod_sofia at 192.168.0.249:5060 >> > RUNNING (0) >> > outbound alias >> external >> > ALIASED >> > 192.168.0.249 alias >> internal >> > ALIASED >> > nat profile >> sip:mod_sofia at 67.171.158.226:5070 >> > RUNNING (0) >> > default alias >> internal >> > ALIASED >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > >> ====================================================================== >> > 3 profiles 3 aliases >> > >> > There is an older thread that says one should set >> > >> > but in this (later) thread is says only Jingleling usese that >> > variable. >> > ie. see: >> > >> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >> org/msg00695.html >> > >> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >> org/msg07345.html >> > >> > So what do you think causes this? What is the correct way? ;-) >> > >> > >> > Thanks, >> > Clif >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> > http://www.freeswitch.org >> >> >> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> http://www.freeswitch.org >> >> >> End of Freeswitch-users Digest, Vol 32, Issue 17 >> ************************************************ >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From wojtek at VoIPMan.ORG Mon Mar 2 04:32:31 2009 From: wojtek at VoIPMan.ORG (Wojciech Tryc) Date: Mon, 2 Mar 2009 07:32:31 -0500 Subject: [Freeswitch-users] Running freeswitch on powerpc In-Reply-To: <7b197bef0903020411q72be83efxace263ab3401c001@mail.gmail.com> References: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> <7b197bef0903020411q72be83efxace263ab3401c001@mail.gmail.com> Message-ID: <60097A1C-2820-4397-BBEE-141FC7FEE3AE@VoIPMan.ORG> Sridhar, PIKA's WARP is PowerPC based...AMCC but still Big Endian and PowerPC. From what I remember the endianness definition was broken in one or two places, but other than that it was effortless (native compilation). Thanks, Wojtek, On Mar 2, 2009, at 7:11 AM, Giovanni Maruzzelli wrote: > On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar) > wrote: >> I am planning to run freeswitch on powerpc MPC8358. Please let me >> know if any changes needs to be done in the code > > Hi Sridhar, > > I don't think someone has tried that. It will probably be you that let > us all know which (if any) changes needs to be done. :-) > > > 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 Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar) > wrote: >> Hi all, >> >> I am planning to run freeswitch on powerpc MPC8358. Please let me >> know if any changes needs to be done in the code >> >> Regards >> Sridhar >> >> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On >>> Behalf Of freeswitch-users-request at lists.freeswitch.org >>> Sent: Monday, February 02, 2009 9:12 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Freeswitch-users Digest, Vol 32, Issue 17 >>> >>> Send Freeswitch-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more >>> specific than "Re: Contents of Freeswitch-users digest..." >>> >>> >>> Today's Topics: >>> >>> 1. Re: Call Variable not available when call hangup (shehzad p) >>> 2. Re: How do I set my FS internal ip address to a "static" >>> value. (clif at eugeneweb.com) >>> 3. Re: Call Variable not available when call hangup >>> (Anthony Minessale) >>> 4. Re: How do I set my FS internal ip address to a "static" >>> value. (Brian West) >>> >>> >>> ---------------------------------------------------------------------- >>> >>> Message: 1 >>> Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST) >>> From: shehzad p >>> Subject: Re: [Freeswitch-users] Call Variable not available when >>> call >>> hangup >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: <21791503.post at talk.nabble.com> >>> Content-Type: text/plain; charset=us-ascii >>> >>> >>> >>> one question is that when javascript is being called from >>> dial plan, I get the session object already available, It is >>> for A leg of channel, So when javascript is called after >>> Bridge how can I get the session object for B leg also? >>> >>> >>> Anthony Minessale-2 wrote: >>>> >>>> the leg you are running the script on is not hungup, the >>> other leg of the >>>> call is. >>>> >>>> If it was hungup you would not be executing the script. >>>> >>>> Asterisk and the h ext and the whole dead-agi thing are all >>> poor design >>>> showing it's teeth. >>>> We do not support anything like it. >>>> >>>> >>>> You can however try this: (see the link below) >>>> >>>> >>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks >>> -giving-me-headaches-p21614840.html >>>> >>>> >>>> >>>> On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: >>>> >>>>> >>>>> Is there any settings that when call hangup control can be >>> transferred to >>>>> another context and these CDR values can be accessible >>> there? (just like >>>>> in >>>>> Asterisk, h extension) >>>>> >>>>> shehzad p wrote: >>>>>> >>>>>> Hi all, >>>>>> >>>>>> I need to process some CDR variables in Dialplan, like >>> call duration, >>>>>> Answered time etc. >>>>>> but when I place info application after bridge, it is >>> not listing them >>>>>> properly as below: >>>>>> =========================================== >>>>>> Caller-Channel-Created-Time: [1233573341672157] >>>>>> Caller-Channel-Answered-Time: [1233573342712939] >>>>>> Caller-Channel-Hangup-Time: [0] >>>>>> ========================================== >>>>>> Here Hangup time is 0, So how can I find actual values? >>>>>> >>>>>> --I know that we can use xml_cdr or cdr_csv, but my >>> current need is to >>>>> get >>>>>> those values from dialplan itself so that can be passed to some >>>>> script... >>>>>> >>>>>> >>>>>> thanks, >>>>>> msp >>>>>> >>>>> >>>>> -- >>>>> View this message in context: >>>>> >>> http://www.nabble.com/Call-Variable-not-available-when-call-ha >>> ngup-tp21788550p21789152.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>> >>>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>> ny.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>> f%2B888 at conference.freeswitch.org> >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Call-Variable-not-available-when-call-ha >>> ngup-tp21788550p21791503.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> >>> >>> ------------------------------ >>> >>> Message: 2 >>> Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST) >>> From: clif at eugeneweb.com >>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip >>> address >>> to a "static" value. >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: >>> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed >>> >>> Hi Gang, >>> >>> I've been struggleing with this also. Actually I can get it >>> to bind to my >>> address, the problem is it randomly drops my calls. :-( >>> >>> I have a FS running on a box with a static IP and I can start >>> a call between >>> two extensions and it will go for hours. Then I add anther >>> interface say eth0:0 >>> with a new static IP and reconfigure my phones and FS to use >>> that, and the >>> calls drop after about 15-20 mins. Though it's pretty random. >>> >>> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and >>> freeswitch-1.0.1. >>> Here is my /etc/network/interfaces: >>> >>> # /etc/network/interfaces -- configuration file for ifup(8), >>> ifdown(8) >>> >>> # The loopback interface >>> auto lo >>> iface lo inet loopback >>> >>> # The first network card - this entry was created during the Debian >>> installation >>> auto eth0 eth0:0 >>> iface eth0 inet dhcp >>> iface eth0:0 inet static >>> address 192.168.0.249 >>> netmask 255.255.255.0 >>> gateway 192.168.0.254 >>> >>> The only change I made to the FS config is in Vars.xml. I >>> added this line close >>> to the top: >>> >>> >>> >>> Here is the console log of the call being dropped: >>> >>> freeswitch at archive> sofia status >>> API CALL [sofia(status)] output: >>> Name Type >>> Data >>> State >>> ============================================================== >>> =================================== >>> external profile >>> sip:mod_sofia at 67.171.158.226:5080 >>> RUNNING (0) >>> internal profile >>> sip:mod_sofia at 192.168.0.249:5060 >>> RUNNING (2) >>> nat profile >>> sip:mod_sofia at 67.171.158.226:5070 >>> RUNNING (0) >>> default alias >>> internal >>> ALIASED >>> outbound alias >>> external >>> ALIASED >>> 192.168.0.249 alias >>> internal >>> ALIASED >>> ============================================================== >>> =================================== >>> 3 profiles 3 aliases >>> >>> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 >>> sofia_glue_restart_all_profiles() Reload XML [Success] >>> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() >>> ENUM Reloaded >>> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 >>> sofia_read_frame() Hangup >>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp >>> ;fs_nat=yes >>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 >>> switch_ivr_multi_threaded_bridge() Hangup >>> sofia/internal/1001 at 192.168.0.249 >>> [CS_EXECUTE] [NORMAL_CLEARING] >>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >>> switch_core_session_thread() Session 6 >>> (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud >>> p;fs_nat=yes) >>> Ended >>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >>> switch_core_session_thread() Close Channel >>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp >>> ;fs_nat=yes >>> [CS_HANGUP] >>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >>> switch_core_session_thread() Session 5 >>> (sofia/internal/1001 at 192.168.0.249) >>> Ended >>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >>> switch_core_session_thread() Close Channel >>> sofia/internal/1001 at 192.168.0.249 >>> [CS_HANGUP] >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() waiting for >>> worker thread >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() waiting for >>> worker thread >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding >>> Alias >>> [192.168.0.249] for profile [internal] >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() >>> Adding Alias [default] >>> for profile [internal] >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() >>> Started Profile >>> internal [sofia_reg_internal] >>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding >>> Alias >>> [outbound] for profile [external] >>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() >>> Started Profile >>> external [sofia_reg_external] >>> 2009-02-01 13:23:20 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() waiting for >>> worker thread >>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() >>> Started Profile nat >>> [sofia_reg_nat] >>> sofia status >>> API CALL [sofia(status)] output: >>> Name Type >>> Data >>> State >>> ============================================================== >>> =================================== >>> external profile >>> sip:mod_sofia at 67.171.158.226:5080 >>> RUNNING (0) >>> internal profile >>> sip:mod_sofia at 192.168.0.249:5060 >>> RUNNING (0) >>> outbound alias >>> external >>> ALIASED >>> 192.168.0.249 alias >>> internal >>> ALIASED >>> nat profile >>> sip:mod_sofia at 67.171.158.226:5070 >>> RUNNING (0) >>> default alias >>> internal >>> ALIASED >>> ============================================================== >>> =================================== >>> 3 profiles 3 aliases >>> >>> There is an older thread that says one should set >>> >>> but in this (later) thread is says only Jingleling usese that >>> variable. >>> ie. see: >>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >>> org/msg00695.html >>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >>> org/msg07345.html >>> >>> So what do you think causes this? What is the correct way? ;-) >>> >>> >>> Thanks, >>> Clif >>> >>> >>> >>> >>> ------------------------------ >>> >>> Message: 3 >>> Date: Mon, 2 Feb 2009 09:41:05 -0600 >>> From: Anthony Minessale >>> Subject: Re: [Freeswitch-users] Call Variable not available when >>> call >>> hangup >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: >>> <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com> >>> Content-Type: text/plain; charset="iso-8859-1" >>> >>> you can't that's why i said it was a horrible approach. >>> That's also why i posted you the instructions on the only >>> elegant solution >>> to your problem. >>> >>> >>> On Mon, Feb 2, 2009 at 9:21 AM, shehzad p wrote: >>> >>>> >>>> >>>> one question is that when javascript is being called from >>> dial plan, I get >>>> the session object already available, It is for A leg of channel, >>>> So when javascript is called after Bridge how can I get the >>> session object >>>> for B leg also? >>>> >>>> >>>> Anthony Minessale-2 wrote: >>>>> >>>>> the leg you are running the script on is not hungup, the >>> other leg of the >>>>> call is. >>>>> >>>>> If it was hungup you would not be executing the script. >>>>> >>>>> Asterisk and the h ext and the whole dead-agi thing are >>> all poor design >>>>> showing it's teeth. >>>>> We do not support anything like it. >>>>> >>>>> >>>>> You can however try this: (see the link below) >>>>> >>>>> >>>> >>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks >>> -giving-me-headaches-p21614840.html >>>>> >>>>> >>>>> >>>>> On Mon, Feb 2, 2009 at 6:53 AM, shehzad p >>> wrote: >>>>> >>>>>> >>>>>> Is there any settings that when call hangup control can >>> be transferred >>>> to >>>>>> another context and these CDR values can be accessible >>> there? (just like >>>>>> in >>>>>> Asterisk, h extension) >>>>>> >>>>>> shehzad p wrote: >>>>>>> >>>>>>> Hi all, >>>>>>> >>>>>>> I need to process some CDR variables in Dialplan, like >>> call duration, >>>>>>> Answered time etc. >>>>>>> but when I place info application after bridge, it is >>> not listing them >>>>>>> properly as below: >>>>>>> =========================================== >>>>>>> Caller-Channel-Created-Time: [1233573341672157] >>>>>>> Caller-Channel-Answered-Time: [1233573342712939] >>>>>>> Caller-Channel-Hangup-Time: [0] >>>>>>> ========================================== >>>>>>> Here Hangup time is 0, So how can I find actual values? >>>>>>> >>>>>>> --I know that we can use xml_cdr or cdr_csv, but my >>> current need is to >>>>>> get >>>>>>> those values from dialplan itself so that can be passed to some >>>>>> script... >>>>>>> >>>>>>> >>>>>>> thanks, >>>>>>> msp >>>>>>> >>>>>> >>>>>> -- >>>>>> View this message in context: >>>>>> >>>> >>> http://www.nabble.com/Call-Variable-not-available-when-call-ha >>> ngup-tp21788550p21789152.html >>>>>> Sent from the Freeswitch-users mailing list archive at >>> Nabble.com. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>> < >>>> >>> MSN%3Aanthony_minessale at hotmail.com>> hotmail.com> >>>>> >>>>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>> ny.minessale at gmail.com> >>>> >>> >> sale at gmail.com> >>>>> >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>> < >>>> >>> sip%3A888 at conference.freeswitch.org>> eswitch.org> >>>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>> googletalk:conf+888 at conference.freeswitch.org>> f%2B888 at conference.freeswitch.org> >>>> >>> >> 253Aconf%252B888 at conference.freeswitch.org> >>>>> >>>>> pstn:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> >>> http://www.nabble.com/Call-Variable-not-available-when-call-ha >>> ngup-tp21788550p21791503.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>> ny.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>> f%2B888 at conference.freeswitch.org> >>> pstn:213-799-1400 >>> -------------- next part -------------- >>> An HTML attachment was scrubbed... >>> URL: >>> http://lists.freeswitch.org/pipermail/freeswitch-users/attachm >>> ents/20090202/2d430e44/attachment-0001.html >>> >>> ------------------------------ >>> >>> Message: 4 >>> Date: Mon, 2 Feb 2009 09:41:39 -0600 >>> From: Brian West >>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip >>> address >>> to a "static" value. >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org> >>> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes >>> >>> you need to add this setting to sofia.conf.xml >>> >>> >>> >>> >>> You'll also need to edit the sofia profiles and input the >>> exact IP you >>> wish it to bind to. The params are sip-ip and rtp-ip. >>> >>> /b >>> >>> On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote: >>> >>>> Hi Gang, >>>> >>>> I've been struggleing with this also. Actually I can get it >>> to bind >>>> to my >>>> address, the problem is it randomly drops my calls. :-( >>>> >>>> I have a FS running on a box with a static IP and I can >>> start a call >>>> between >>>> two extensions and it will go for hours. Then I add anther >>> interface >>>> say eth0:0 >>>> with a new static IP and reconfigure my phones and FS to use that, >>>> and the >>>> calls drop after about 15-20 mins. Though it's pretty random. >>>> >>>> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and >>>> freeswitch-1.0.1. >>>> Here is my /etc/network/interfaces: >>>> >>>> # /etc/network/interfaces -- configuration file for >>> ifup(8), ifdown(8) >>>> >>>> # The loopback interface >>>> auto lo >>>> iface lo inet loopback >>>> >>>> # The first network card - this entry was created during the Debian >>>> installation >>>> auto eth0 eth0:0 >>>> iface eth0 inet dhcp >>>> iface eth0:0 inet static >>>> address 192.168.0.249 >>>> netmask 255.255.255.0 >>>> gateway 192.168.0.254 >>>> >>>> The only change I made to the FS config is in Vars.xml. I >>> added this >>>> line close >>>> to the top: >>>> >>>> >>>> >>>> Here is the console log of the call being dropped: >>>> >>>> freeswitch at archive> sofia status >>>> API CALL [sofia(status)] output: >>>> Name Type >>> Data >>>> State >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> >>> = >>> = >>> ==================================================================== >>>> external profile >>> sip:mod_sofia at 67.171.158.226:5080 >>>> RUNNING (0) >>>> internal profile >>> sip:mod_sofia at 192.168.0.249:5060 >>>> RUNNING (2) >>>> nat profile >>> sip:mod_sofia at 67.171.158.226:5070 >>>> RUNNING (0) >>>> default alias >>> internal >>>> ALIASED >>>> outbound alias >>> external >>>> ALIASED >>>> 192.168.0.249 alias >>> internal >>>> ALIASED >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> >>> = >>> = >>> ==================================================================== >>>> 3 profiles 3 aliases >>>> >>>> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 >>>> sofia_glue_restart_all_profiles() Reload XML [Success] >>>> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM >>>> Reloaded >>>> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 >>> sofia_read_frame() Hangup >>>> sofia/internal/ >>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes >>>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 >>>> switch_ivr_multi_threaded_bridge() Hangup >>> sofia/internal/1001 at 192.168.0.249 >>>> [CS_EXECUTE] [NORMAL_CLEARING] >>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >>>> switch_core_session_thread() Session 6 >>>> (sofia/internal/ >>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) >>>> Ended >>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >>>> switch_core_session_thread() Close Channel >>>> sofia/internal/ >>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes >>>> [CS_HANGUP] >>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >>>> switch_core_session_thread() Session 5 >>> (sofia/internal/1001 at 192.168.0.249 >>>> ) >>>> Ended >>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >>>> switch_core_session_thread() Close Channel >>> sofia/internal/1001 at 192.168.0.249 >>>> [CS_HANGUP] >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() >>>> waiting for >>>> worker thread >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() >>>> waiting for >>>> worker thread >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() >>> Adding Alias >>>> [192.168.0.249] for profile [internal] >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding >>>> Alias [default] >>>> for profile [internal] >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started >>>> Profile >>>> internal [sofia_reg_internal] >>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() >>> Adding Alias >>>> [outbound] for profile [external] >>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started >>>> Profile >>>> external [sofia_reg_external] >>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() >>>> waiting for >>>> worker thread >>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started >>>> Profile nat >>>> [sofia_reg_nat] >>>> sofia status >>>> API CALL [sofia(status)] output: >>>> Name Type >>> Data >>>> State >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> >>> = >>> = >>> ==================================================================== >>>> external profile >>> sip:mod_sofia at 67.171.158.226:5080 >>>> RUNNING (0) >>>> internal profile >>> sip:mod_sofia at 192.168.0.249:5060 >>>> RUNNING (0) >>>> outbound alias >>> external >>>> ALIASED >>>> 192.168.0.249 alias >>> internal >>>> ALIASED >>>> nat profile >>> sip:mod_sofia at 67.171.158.226:5070 >>>> RUNNING (0) >>>> default alias >>> internal >>>> ALIASED >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> >>> = >>> = >>> ==================================================================== >>>> 3 profiles 3 aliases >>>> >>>> There is an older thread that says one should set >>>> >>>> but in this (later) thread is says only Jingleling usese that >>>> variable. >>>> ie. see: >>>> >>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >>> org/msg00695.html >>>> >>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >>> org/msg07345.html >>>> >>>> So what do you think causes this? What is the correct way? ;-) >>>> >>>> >>>> Thanks, >>>> Clif >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>> http://www.freeswitch.org >>> >>> >>> End of Freeswitch-users Digest, Vol 32, Issue 17 >>> ************************************************ >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveu at coppice.org Mon Mar 2 04:50:29 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 02 Mar 2009 20:50:29 +0800 Subject: [Freeswitch-users] Running freeswitch on powerpc In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <49ABD615.9050906@coppice.org> Rajagopal, Sridhar (Sridhar) wrote: > Hi all, > > I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code > > Regards > Sridhar > It may be easier to say what will currently stop Freeswitch working. The lack of an MMU is a problem right now, so Blackfins are out, which is sad. Cores without hardware floating point may not perform all that well, but should work. Endianness should not be a problem. I think machines which choke on misaligned access are probably OK, too. Checking that list, you should be OK on a PPC. Steve From anthony.minessale at gmail.com Mon Mar 2 05:50:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 07:50:32 -0600 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <2ea4d47e0903020129y676bbc35n8727f8906237edce@mail.gmail.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <49A6E1DB.3070806@freeswitch.org> <1235740392995-2395557.post@n2.nabble.com> <2ea4d47e0903020129y676bbc35n8727f8906237edce@mail.gmail.com> Message-ID: <191c3a030903020550x44ee80e3tcf33b805c7c30d5e@mail.gmail.com> pardon? ESL is just a client library for event socket to make it easier to make event socket apps. ESL == Event Socket Library On Mon, Mar 2, 2009 at 3:29 AM, Gopal krishnan wrote: > Hi, > Actually what is the difference between ESL in FS 1.0.3 and event socket > in FS 1.0.2. Is the FS 1.0.3 ESL superior? > > On Fri, Feb 27, 2009 at 6:43 PM, Rex_Alex wrote: > >> Hi All, I did what you have all suggested. Now its working perfectly. >> Thanks a lot for all your assistance. Rex. >> >> Raymond Chandler wrote: >> and it will probably be a good idea to do make phpmod-install so that the >> .so and .php files gets into the correct place to be included -Ray Mathieu >> Rene wrote: > > You need your distro's php dev pakage. > On 26-Feb-09, at >> 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh && >> ./configure && make >> install and did Mathieu's suggestion but getting >> error as below, >> [root at server esl]# make phpmod make >> MYLIB="../libesl.a" >> SOLINK="-shared -Xlinker -x" >> >> CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >> -g >> -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >> -Werror >> -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> >> -Wmissing-prototypes" >> >> CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >> -DHAVE_EDITLINE >> -g -ggdb -I../../libs/libedit/src/ -fPIC" >> CXX_CFLAGS="" -C php make[1]: >> php-config: Command not found make[1]: >> Entering directory >> `/root/freeswitch-1.0.3/libs/esl/php' g++ >> >> -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >> -ggdb >> -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >> >> esl_wrap.cpp:717:18: error: zend.h: No such file or directory >> >> esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >> >> esl_wrap.cpp:719:17: error: php.h: No such file or directory >> >> esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >> >> esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >> >> directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >> >> scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >> >> ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >> >> ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >> error: >> expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >> variable or >> field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error: >> ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was >> not declared in this scope >> esl_wrap.cpp:793: error: expected >> primary-expression before ?void? >> esl_wrap.cpp:793: error: expected >> primary-expression before ?*? token >> esl_wrap.cpp:793: error: ?type? was >> not declared in this scope >> esl_wrap.cpp:793: error: expected >> primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer >> expression list treated as >> compound expression esl_wrap.cpp:793: error: >> expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 >> make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: >> *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i >> wrong? Thanks, Rex >> >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34 >> AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it.. >> Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh && >> ./configure && >> make install And then do Mathieu's suggestion? -MC >> >> _______________________________________________ Freeswitch-users >> mailing >> list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>> >> http://www.freeswitch.org >> >> >> >> ------------------------------------------------------------------------ >> >> View this message in context: Re: ESL Wrapper >> >> Sent from the >> freeswitch-users mailing list archive >> at Nabble.com. >> >> _______________________________________________ >> Freeswitch-users mailing >> list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> View this message in context: Re: ESL Wrapper >> Sent from the freeswitch-users mailing list archiveat Nabble.com. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thank you with regards, > Gopal, > PeopleTech Systems Private Limited > www.peopletech.co.in > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/0bf9dec0/attachment.html From anthony.minessale at gmail.com Mon Mar 2 05:56:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 07:56:50 -0600 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <49AB004C.6090604@gmx.net> References: <49AB004C.6090604@gmx.net> Message-ID: <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> put origination_caller_id_number in the dial string of any call and you can set the caller id individually for that leg {origination_caller_id_number=1234} On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX wrote: > Hello, > > I have the following problem while providing callback (mod_eventsocket > is used): > 1) I want to call a certain destination number A with a suppressed > caller_id_number (this works fine with some vars in the origination string) > 2) The destination number A picks up the phone and enters a target > number B by DTMF > 3) freeswitch then forwards the call to target number B by DTMF and I > want to show the number A. I do this with uuid_setvar. The problem is > that it still shows unknown. > This is all with SIP. > > uuid_setvar however worked if I did not set the caller_id_number to > unknown. Per default this is then "00000000000" and can then be changed > with uuid_setvar to the number of A. > But if I set caller_id_number to unknown I can no longer change it to A. > > Any hint? > > Best regards > Peter > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/7d5ce07e/attachment.html From anthony.minessale at gmail.com Mon Mar 2 05:59:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 07:59:34 -0600 Subject: [Freeswitch-users] Orginate: getting status of call fail In-Reply-To: References: Message-ID: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com> The best way would be to add a few custom variables and add a secondary system that monitors the CDR data and uses the custom variables to identify what you want to do with the failed calls. On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I?ve been running a test script written in lua which now works very well > thanks to Anthony?s fix to stream file. > > > > Right now I?m using an event socket to initiate the call and passing the > name of the script along with originate thus: > > > > $dialstring = "originate > {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/Mygw/phonenum > '&lua(helloworld.lua )'"; > > $result = $obj ->bgapi_command($dialstring); > > > > The script gets fired (it would appear) on answer. However, if the number > is invalid , timed out or was busy, I?m not sure the script gets executed or > am I wrong? > > > > I want to be able to fire an event back on what happed to the call in the > event that it failed for whatever reason. > > > > I know I can simply call the originate and pass the number as an argument > and execute the dial within the script but I?m led to believe that?s not > very efficient, or am I completely wrong? > > > > Looking for the most FS friendly way here > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/d35279cc/attachment-0001.html From nik.middleton at noblesolutions.co.uk Mon Mar 2 06:49:03 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 2 Mar 2009 14:49:03 -0000 Subject: [Freeswitch-users] Orginate: getting status of call fail In-Reply-To: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com> References: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com> Message-ID: That's what I was wondering, however, won't the response to the bagi (not the initial) give me the info on the call result? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 02 March 2009 14:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orginate: getting status of call fail The best way would be to add a few custom variables and add a secondary system that monitors the CDR data and uses the custom variables to identify what you want to do with the failed calls. On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton wrote: Hi Guys, I've been running a test script written in lua which now works very well thanks to Anthony's fix to stream file. Right now I'm using an event socket to initiate the call and passing the name of the script along with originate thus: $dialstring = "originate {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/ Mygw/phonenum '&lua(helloworld.lua )'"; $result = $obj ->bgapi_command($dialstring); The script gets fired (it would appear) on answer. However, if the number is invalid , timed out or was busy, I'm not sure the script gets executed or am I wrong? I want to be able to fire an event back on what happed to the call in the event that it failed for whatever reason. I know I can simply call the originate and pass the number as an argument and execute the dial within the script but I'm led to believe that's not very efficient, or am I completely wrong? Looking for the most FS friendly way here Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/2595216a/attachment.html From anthony.minessale at gmail.com Mon Mar 2 07:26:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 09:26:57 -0600 Subject: [Freeswitch-users] Orginate: getting status of call fail In-Reply-To: References: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com> Message-ID: <191c3a030903020726y455b786dka15206f2be5a7559@mail.gmail.com> yes if you match the job uuid from bgapi to the SWITCH_EVENT_BACKGROUND_JOB event, you would get the result in that event. On Mon, Mar 2, 2009 at 8:49 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > That?s what I was wondering, however, won?t the response to the bagi (not > the initial) give me the info on the call result? > > > > Regards > > > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 02 March 2009 14:00 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Orginate: getting status of call fail > > > > The best way would be to add a few custom variables and add a secondary > system that monitors the CDR data and uses the > custom variables to identify what you want to do with the failed calls. > > > On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Hi Guys, > > > > I?ve been running a test script written in lua which now works very well > thanks to Anthony?s fix to stream file. > > > > Right now I?m using an event socket to initiate the call and passing the > name of the script along with originate thus: > > > > $dialstring = "originate > {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/Mygw/phonenum > '&lua(helloworld.lua )'"; > > $result = $obj ->bgapi_command($dialstring); > > > > The script gets fired (it would appear) on answer. However, if the number > is invalid , timed out or was busy, I?m not sure the script gets executed or > am I wrong? > > > > I want to be able to fire an event back on what happed to the call in the > event that it failed for whatever reason. > > > > I know I can simply call the originate and pass the number as an argument > and execute the dial within the script but I?m led to believe that?s not > very efficient, or am I completely wrong? > > > > Looking for the most FS friendly way here > > > > Regards, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/f3e9cdd0/attachment.html From a.playful.idiot at gmail.com Sun Mar 1 23:43:34 2009 From: a.playful.idiot at gmail.com (Aplayful Idiot) Date: Sun, 1 Mar 2009 23:43:34 -0800 Subject: [Freeswitch-users] First time test set up FreeSwitch and SPA3102/SPA3000 Message-ID: <9ed22e920903012343u5e50c89fkfa9d1127894199c@mail.gmail.com> I have no background in telephony but probably need to use a PBX. FreeSwitch was recommended by a casual contact so I would like to start first by setting up a small test. I have a SPA3102 attached to the box running FS and to a ordinary phone line. I registered SPA in conf/directory/default/line1.xml and it works ok but I can't get caller id numbers from incoming calls. All FS sees is "line1" which is found in file line1.xml as . Looking back over the FS wiki, I'm now wondering if the SPA should of been set-up as a gateway but reading it is confusing at least to me. Sometimes I think the analogue-line-SPA-FS is like a softphone which is registered to an extension numbered xml file in conf/directory/default/ but then issues like not getting outside incoming caller id's makes me think I've got this all wrong. Can someone help me out with this? api -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/eb6ea6e5/attachment-0001.html From a.playful.idiot at gmail.com Mon Mar 2 07:41:31 2009 From: a.playful.idiot at gmail.com (Aplayful Idiot) Date: Mon, 2 Mar 2009 07:41:31 -0800 Subject: [Freeswitch-users] First time setting up FreeSwitch and SPA3102 / SPA3000 Message-ID: <9ed22e920903020741o2ff66970h35d3182655b2c7ba@mail.gmail.com> Hi. I have little background in telephony and need to use a PBX but would like to start first with a small test set-up. I have a SPA3102 attached to the box running FS and to a ordinary phone line. I registered SPA in conf/directory/default/line1.xml and this works to a point but I can't get caller id numbers from incoming calls. All FS sees is "line1" which is found in file line1.xml as . Looking back over the FS wiki, I'm now wondering if the SPA was registered or set-up in FS correctly but reading the documentation is confusing me a bit. Sometimes I think the analogue-phone-line-SPA-FS is like a softphone which is registered to an extension numbered xml file in conf/directory/default/ but then issues like not getting outside incoming caller id's makes me think I've got this all wrong. Can someone help me out with this? Thanks. api -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/e073e954/attachment.html From mike at jerris.com Mon Mar 2 08:14:52 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Mar 2009 11:14:52 -0500 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: <49AB8562.4050806@laposte.net> References: <49AB8562.4050806@laposte.net> Message-ID: Could you please post this to jira along with a thread apply all bt of a core file taken from the process with the stuck sessions. Mike On Mar 2, 2009, at 2:06 AM, rod wrote: > Hi All, > > I ran some longer tests with FS 1.0.3 acting as an SBC. > The test machine has the following specs: > - Intel Quad Core Q9550 > - 8GB RAM (far too much from what I saw) > > After 3 days running SIPP with 750 simultaneous calls (1500 > channels) at > 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. > > In the CLI, using status command I got this: > > freeswitch at internal> status > UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds, > 607 microseconds > 15817560 session(s) since startup > 22 session(s) 0/500 > > But when I use "show channels" or "show calls", I see nothing. So I'm > wondering where are these 22 sessions ? > > FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs. > > Successful call --> 5271434 > Failed call ---> 1554 (less than 0.03%) > > regards, > rod. > > > > complete SIPP summary: > > ------------------------------ Scenario Screen -------- [1-9]: Change > Screen -- > Call-rate(length) Port Total-time Total-calls Remote-host > 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 > 10.10.10.254:5060(UDP) > > 0 new calls during 0.856 s period 7 ms scheduler resolution > 0 calls (limit 750) Peak was 750 calls, after 15 s > 0 Running, 34 Paused, 0 Woken up > 15544 out-of-call msg (discarded) > 1 open sockets > 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate > (kB/s) > 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate > (kB/s) > > Messages Retrans Timeout > Unexpected-Msg > INVITE ----------> 5273022 0 0 > 100 <---------- 5273022 0 1554 > 180 <---------- 0 0 0 > 183 <---------- 0 0 0 > 200 <---------- E-RTD1 5271434 0 0 > ACK ----------> 5271434 0 > Pause [ 35.0s] 5271434 0 > BYE ----------> 5271434 0 0 > 200 <---------- 5271434 0 0 > > ------------------------------ Test Terminated > -------------------------------- > > > ----------------------------- Statistics Screen ------- [1-9]: Change > Screen -- > Start Time | 2009-02-27 > 09:11:31 > Last Reset Time | 2009-03-02 > 07:49:10 > Current Time | 2009-03-02 > 07:49:11 > -------------------------+--------------------------- > +-------------------------- > Counter Name | Periodic value | Cumulative value > -------------------------+--------------------------- > +-------------------------- > Elapsed Time | 00:00:00:857 | > 70:37:39:429 > Call Rate | 0.000 cps | 20.739 > cps > -------------------------+--------------------------- > +-------------------------- > Incoming call created | 0 | > 0 > OutGoing call created | 0 | > 5273022 > Total Call created | | > 5273022 > Current Call | 34 > | > -------------------------+--------------------------- > +-------------------------- > Successful call | 0 | > 5271434 > Failed call | 0 | > 1554 > -------------------------+--------------------------- > +-------------------------- > Response Time 1 | 00:00:00:000 | > 00:00:00:240 > Call Length | 38:32:13:386 | > 00:00:36:131 > ------------------------------ Test Terminated > -------------------------------- From mike at jerris.com Mon Mar 2 08:18:42 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Mar 2009 11:18:42 -0500 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? In-Reply-To: <7.0.1.0.2.20090302124612.028657a8@free.fr> References: <7.0.1.0.2.20090302124612.028657a8@free.fr> Message-ID: <485E559F-4D98-48AB-8CF0-EF3FD50DBB5F@jerris.com> I think any issues we have are related to pri, the analog doesn't seem to generate any major bug reports. Mike On Mar 2, 2009, at 6:47 AM, Fred wrote: > Thanks guys for the feedback. So, the OpenZap driver isn't ready for > production yet? > From Prometheus001 at gmx.net Mon Mar 2 08:19:32 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 02 Mar 2009 17:19:32 +0100 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> Message-ID: <49AC0714.8000006@gmx.net> Hello Anthony, I do this when I orginate the call. This way we suppress the cid when we call party A and transfer A to an internal extension (our callback application). But now comes the part that does not work: After A enters the target number B (via DTMF), we set the cid variables via uuid_setvar and then transfer A via uuid_transfer to party B. However uuid_setvar does not work in that case. BUT: If we do the same scenario and do not suppress the cid in the originate part, then uuid_setvar works correctly and sets the cid_number. Best regards Peter Anthony Minessale schrieb: > put origination_caller_id_number in the dial string of any call and > you can set the caller id individually for that leg > > {origination_caller_id_number=1234} > > > On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX > wrote: > > Hello, > > I have the following problem while providing callback (mod_eventsocket > is used): > 1) I want to call a certain destination number A with a suppressed > caller_id_number (this works fine with some vars in the > origination string) > 2) The destination number A picks up the phone and enters a target > number B by DTMF > 3) freeswitch then forwards the call to target number B by DTMF and I > want to show the number A. I do this with uuid_setvar. The problem is > that it still shows unknown. > This is all with SIP. > > uuid_setvar however worked if I did not set the caller_id_number to > unknown. Per default this is then "00000000000" and can then be > changed > with uuid_setvar to the number of A. > But if I set caller_id_number to unknown I can no longer change it > to A. > > Any hint? > > Best regards > Peter > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From woof at nortel.com Mon Mar 2 09:16:33 2009 From: woof at nortel.com (Andy Spitzer) Date: Mon, 02 Mar 2009 12:16:33 -0500 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: Woof! On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote: > NO. You want something that people THINK exists and works well... > Reliable human/voice detection doesn't exist in ANY form. I beg to differ. See http://www.freepatentsonline.com/5521967.html for one way to do it. It works rather well and can quickly descriminate between voice and tone. I've no idea who owns that patent now (not me, for sure). There is a simpler, less reliable way of differentiating voice from tone, that as far as I know isn't patented. If you compare the RMS power levels of sequential 40 mS periods, call progress tones will have very consistent power levels from sample to sample. So if 5 or more 40 mS periods have about the same power measurement (within say, 2%), it's a tone. Voice will have dramatic power level differences over that same period. This works very well in today's telephony environment, where tones are computer generated. In the old days when ringback tone was generated off the audio hum from the 20 Hz ring voltage generator...not so well. --Woof! From msc at freeswitch.org Mon Mar 2 10:05:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Mar 2009 10:05:43 -0800 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49A7F393.6080406@ewetel.de> References: <49A7F393.6080406@ewetel.de> Message-ID: <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> On Fri, Feb 27, 2009 at 6:07 AM, Helmut Kuper wrote: > Hello, > > I play around with record_session and would like to have caller and > callee separated on left and right channel. I found record_stereo is > used for this. Unfortunately it doesn't work. A and B leg are still > mixed. Additionally I found that B leg is significant louder than A leg, > but both legs were local extensions. > Just to confirm - you are trying to record each leg of the call into a separate file? In other words, one call creates two separate audio recordings? -MC From anthony.minessale at gmail.com Mon Mar 2 11:48:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 13:48:26 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> i think that's what mod_vmd does On Mon, Mar 2, 2009 at 11:16 AM, Andy Spitzer wrote: > Woof! > > On Sun, 01 Mar 2009 21:28:18 -0500, Brian West > wrote: > > > NO. You want something that people THINK exists and works well... > > Reliable human/voice detection doesn't exist in ANY form. > > I beg to differ. See http://www.freepatentsonline.com/5521967.html for > one way to do it. It works rather well and can quickly descriminate between > voice and tone. I've no idea who owns that patent now (not me, for sure). > > There is a simpler, less reliable way of differentiating voice from tone, > that as far as I know isn't patented. If you compare the RMS power levels > of sequential 40 mS periods, call progress tones will have very consistent > power levels from sample to sample. So if 5 or more 40 mS periods have > about the same power measurement (within say, 2%), it's a tone. Voice will > have dramatic power level differences over that same period. This works > very well in today's telephony environment, where tones are computer > generated. In the old days when ringback tone was generated off the audio > hum from the 20 Hz ring voltage generator...not so well. > > --Woof! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/a8783b8b/attachment.html From anthony.minessale at gmail.com Mon Mar 2 11:52:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 13:52:16 -0600 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <49AC0714.8000006@gmx.net> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> <49AC0714.8000006@gmx.net> Message-ID: <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> origination_caller_id number is not ok as a variable unless its in {} as part of the dial string it's an exception that is parsed before the channel is even created. I think you are drawing the wrong conclusion about what works and doesn't work. If you can produce a dial string that contains {origination_caller_id_number=x} you will always be able to set it. I assume you are using a recent version of FS as we did have a small bug with this variable a few weeks ago. On Mon, Mar 2, 2009 at 10:19 AM, Peter P GMX wrote: > Hello Anthony, > > I do this when I orginate the call. This way we suppress the cid when we > call party A and transfer A to an internal extension (our callback > application). > But now comes the part that does not work: > After A enters the target number B (via DTMF), we set the cid variables > via uuid_setvar and then transfer A via uuid_transfer to party B. > However uuid_setvar does not work in that case. > > BUT: If we do the same scenario and do not suppress the cid in the > originate part, then uuid_setvar works correctly and sets the cid_number. > > Best regards > Peter > > Anthony Minessale schrieb: > > put origination_caller_id_number in the dial string of any call and > > you can set the caller id individually for that leg > > > > {origination_caller_id_number=1234} > > > > > > On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX > > wrote: > > > > Hello, > > > > I have the following problem while providing callback > (mod_eventsocket > > is used): > > 1) I want to call a certain destination number A with a suppressed > > caller_id_number (this works fine with some vars in the > > origination string) > > 2) The destination number A picks up the phone and enters a target > > number B by DTMF > > 3) freeswitch then forwards the call to target number B by DTMF and I > > want to show the number A. I do this with uuid_setvar. The problem is > > that it still shows unknown. > > This is all with SIP. > > > > uuid_setvar however worked if I did not set the caller_id_number to > > unknown. Per default this is then "00000000000" and can then be > > changed > > with uuid_setvar to the number of A. > > But if I set caller_id_number to unknown I can no longer change it > > to A. > > > > Any hint? > > > > Best regards > > Peter > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/21522899/attachment-0001.html From msc at freeswitch.org Mon Mar 2 12:03:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Mar 2009 12:03:52 -0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> Message-ID: <87f2f3b90903021203y3588ffa4nc3ccd280117e0129@mail.gmail.com> On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale wrote: > i think that's what mod_vmd does > I think that's right. It just does the opposite - instead of looking for differing power levels it looks for the same power level. In other words it tries to detect distinctly non-human sound. I'll bet you could futz with that code and have it fire off events when it detects what it believes is human speech. -MC From Prometheus001 at gmx.net Mon Mar 2 12:31:47 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 02 Mar 2009 21:31:47 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AB0BCD.8030108@gmx.net> References: <49A92BAE.4090907@gmx.net> <49AB0BCD.8030108@gmx.net> Message-ID: <49AC4233.6060506@gmx.net> Some more info: the system I am working on is a copy (dd copy) of a system where the pizza demo works on. The only thing I changed was to update to the current freeswitch trunk 12293 (it was 10003 before). Do I need to update the model? I did a make in the model directory, but no change. Best regards Peter Peter P GMX schrieb: > Hello Brian, > > thanks for the info. I am a step further, but it cannot load the grammar > files. > I am sending through event_socket: > > SendMsg > call-command: execute > execute-app-name: detect_speech > execute-app-arg: pocketsphinx yes no > > However I get the message (also when I am using Pizza demo): > 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() > sofia/internal/1000 at sip2.server.com Command Execute > detect_speech(pocketsphinx yes no) > 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 > pocketsphinx_asr_load_grammar() Can't open language model > /usr/local/freeswitch/grammar/model/communicator. > 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 > switch_ivr_detect_speech() Error loading Grammar > 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 > pocketsphinx_asr_close() Port Closed. > > However the grammar files are there: > root at sip2:/usr/local/freeswitch/grammar/model/communicator# > root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al > total 12752 > drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . > drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. > -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING > -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params > -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef > -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means > -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict > -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump > -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices > -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances > > > Any hint? > > Best regards > Peter > > Brian West schrieb: > >> You can accomplish this .... here is an example using ESL in perl >> >> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 >> >> /b >> >> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: >> >> >> >>> Or back to the basics: Is it possible to use pocketsphinx through >>> event >>> socket? >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From sergio.alecha at gmail.com Mon Mar 2 12:18:06 2009 From: sergio.alecha at gmail.com (Sergio Alecha) Date: Mon, 2 Mar 2009 17:18:06 -0300 Subject: [Freeswitch-users] Howto config early dial Message-ID: <47f9a0940903021218l1a6f7ea1l8c596ddd3514e86c@mail.gmail.com> In asterisk, with the parameter AMPBADNUMBER = FALSE it is possible to use "early dial" Grandstream telephones. How do Freeswitch in? thank you very much. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/beb261ca/attachment.html From Prometheus001 at gmx.net Mon Mar 2 13:58:35 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 02 Mar 2009 22:58:35 +0100 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> <49AC0714.8000006@gmx.net> <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> Message-ID: <49AC568B.7020504@gmx.net> Hello Anthony, sorry for being tenacious but in some cases it works in a way we need it: If I a am not suppressing the cid numer when calling A, the following scenario works: * A receives a Call (originate) with CID '0000000000' (default from switch_caller.c) * A dials some digits via DTMF, the app set the cid variables via uuid_setvar and uuid_transfers the call to B. B receives a call with the right cid set. Maybe I simply modify the default cid '0000000000' to a different value in switch_caller.c? Is there a special reason why this is '0000000000'? I am using trunk version 12293. Best regards Peter Anthony Minessale schrieb: > origination_caller_id number is not ok as a variable unless its in {} > as part of the dial string > it's an exception that is parsed before the channel is even created. > > I think you are drawing the wrong conclusion about what works and > doesn't work. > If you can produce a dial string that contains > {origination_caller_id_number=x} you will always be able to set it. > > I assume you are using a recent version of FS as we did have a small > bug with this variable a few weeks ago. > > > On Mon, Mar 2, 2009 at 10:19 AM, Peter P GMX > wrote: > > Hello Anthony, > > I do this when I orginate the call. This way we suppress the cid > when we > call party A and transfer A to an internal extension (our callback > application). > But now comes the part that does not work: > After A enters the target number B (via DTMF), we set the cid > variables > via uuid_setvar and then transfer A via uuid_transfer to party B. > However uuid_setvar does not work in that case. > > BUT: If we do the same scenario and do not suppress the cid in the > originate part, then uuid_setvar works correctly and sets the > cid_number. > > Best regards > Peter > > Anthony Minessale schrieb: > > put origination_caller_id_number in the dial string of any call and > > you can set the caller id individually for that leg > > > > {origination_caller_id_number=1234} > > > > > > On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX > > > >> > wrote: > > > > Hello, > > > > I have the following problem while providing callback > (mod_eventsocket > > is used): > > 1) I want to call a certain destination number A with a > suppressed > > caller_id_number (this works fine with some vars in the > > origination string) > > 2) The destination number A picks up the phone and enters a > target > > number B by DTMF > > 3) freeswitch then forwards the call to target number B by > DTMF and I > > want to show the number A. I do this with uuid_setvar. The > problem is > > that it still shows unknown. > > This is all with SIP. > > > > uuid_setvar however worked if I did not set the > caller_id_number to > > unknown. Per default this is then "00000000000" and can then be > > changed > > with uuid_setvar to the number of A. > > But if I set caller_id_number to unknown I can no longer > change it > > to A. > > > > Any hint? > > > > Best regards > > Peter > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Mar 2 14:08:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Mar 2009 14:08:53 -0800 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <49AC568B.7020504@gmx.net> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> <49AC0714.8000006@gmx.net> <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> <49AC568B.7020504@gmx.net> Message-ID: <87f2f3b90903021408g62ba6a4eo925b70beb1a562a5@mail.gmail.com> On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX wrote: > Hello Anthony, > > sorry for being tenacious but in some cases it works in a way we need it: > If I a am not suppressing the cid numer when calling A, the following > scenario works: > > ? ?* A receives a Call (originate) with CID '0000000000' (default from > ? ? ?switch_caller.c) > ? ?* A dials some digits via DTMF, the app set the cid variables via > ? ? ?uuid_setvar and uuid_transfers the call to B. B receives a call > ? ? ?with the right cid set. > > Maybe I simply modify the default cid '0000000000' ?to a different value > in switch_caller.c? Is there a special reason why this is '0000000000'? > Check vars.xml to confirm that you have actually set a default caller ID. Most likely you'll still have the default caller id number set to all zeroes, which is the default. -MC > I am using trunk version 12293. > > Best regards > Peter > From anthony.minessale at gmail.com Mon Mar 2 14:22:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 16:22:25 -0600 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <87f2f3b90903021408g62ba6a4eo925b70beb1a562a5@mail.gmail.com> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> <49AC0714.8000006@gmx.net> <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> <49AC568B.7020504@gmx.net> <87f2f3b90903021408g62ba6a4eo925b70beb1a562a5@mail.gmail.com> Message-ID: <191c3a030903021422i385ab5c0h781993c65d6d27b4@mail.gmail.com> Since you did not describe the exact way you are doing it with enough detail or any trace I can't begin to tell you what your problem is. you did not even mention what variable you are using or show examples. All I can do is tell you again that if you set the origination_caller_id_number in the dial string it will be the most likely to work for you. On Mon, Mar 2, 2009 at 4:08 PM, Michael Collins wrote: > On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX wrote: > > Hello Anthony, > > > > sorry for being tenacious but in some cases it works in a way we need it: > > If I a am not suppressing the cid numer when calling A, the following > > scenario works: > > > > * A receives a Call (originate) with CID '0000000000' (default from > > switch_caller.c) > > * A dials some digits via DTMF, the app set the cid variables via > > uuid_setvar and uuid_transfers the call to B. B receives a call > > with the right cid set. > > > > Maybe I simply modify the default cid '0000000000' to a different value > > in switch_caller.c? Is there a special reason why this is '0000000000'? > > > > Check vars.xml to confirm that you have actually set a default caller > ID. Most likely you'll still have the default caller id number set to > all zeroes, which is the default. > > -MC > > > I am using trunk version 12293. > > > > Best regards > > Peter > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/58b8ff69/attachment.html From freeswitch at servercorps.com Mon Mar 2 14:43:04 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Mon, 2 Mar 2009 16:43:04 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AC4233.6060506@gmx.net> References: <49A92BAE.4090907@gmx.net> <49AB0BCD.8030108@gmx.net> <49AC4233.6060506@gmx.net> Message-ID: <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com> Peter, You need the grammar files for the pizza demo: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo has lonks to premade fles for everyhting to get the pizza demo working with pocketshinx. Those to not come with the source code when you update from SVN. Nik On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX wrote: > Some more info: > the system I am working on is a copy (dd copy) of a system where the > pizza demo works on. > The only thing I changed was to update to the current freeswitch trunk > 12293 (it was 10003 before). > > Do I need to update the model? I did a make in the model directory, but > no change. > > Best regards > Peter > > Peter P GMX schrieb: >> Hello Brian, >> >> thanks for the info. I am a step further, but it cannot load the grammar >> files. >> I am sending through event_socket: >> >> SendMsg >> call-command: execute >> execute-app-name: detect_speech >> execute-app-arg: pocketsphinx yes no >> >> However I get the message (also when I am using Pizza demo): >> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() >> sofia/internal/1000 at sip2.server.com Command Execute >> detect_speech(pocketsphinx yes no) >> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 >> pocketsphinx_asr_load_grammar() Can't open language model >> /usr/local/freeswitch/grammar/model/communicator. >> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 >> switch_ivr_detect_speech() Error loading Grammar >> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 >> pocketsphinx_asr_close() Port Closed. >> >> However the grammar files are there: >> root at sip2:/usr/local/freeswitch/grammar/model/communicator# >> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al >> total 12752 >> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . >> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. >> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING >> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params >> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef >> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means >> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict >> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump >> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices >> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances >> >> >> Any hint? >> >> Best regards >> Peter >> >> Brian West schrieb: >> >>> You can accomplish this .... here is an example using ESL in perl >>> >>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 >>> >>> /b >>> >>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: >>> >>> >>> >>>> Or back to the basics: Is it possible to use pocketsphinx through >>>> event >>>> socket? >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Mon Mar 2 15:42:28 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 03 Mar 2009 00:42:28 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com> References: <49A92BAE.4090907@gmx.net> <49AB0BCD.8030108@gmx.net> <49AC4233.6060506@gmx.net> <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com> Message-ID: <49AC6EE4.9080509@gmx.net> Thanks Addison. The Pizza files are there (as mentionned is it a copy of an already working system). In fact freeswitch is complaning about /usr/local/freeswitch/grammar/model/communicator which he cannot load So somehow freeswitch is not willing to open the files, but I have no clue why. So any hints are welcome. Best regards Peter Addison Martin schrieb: > Peter, > > You need the grammar files for the pizza demo: > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo > has lonks to premade fles for everyhting to get the pizza demo working > with pocketshinx. Those to not come with the source code when you > update from SVN. > > Nik > > > > On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX wrote: > >> Some more info: >> the system I am working on is a copy (dd copy) of a system where the >> pizza demo works on. >> The only thing I changed was to update to the current freeswitch trunk >> 12293 (it was 10003 before). >> >> Do I need to update the model? I did a make in the model directory, but >> no change. >> >> Best regards >> Peter >> >> Peter P GMX schrieb: >> >>> Hello Brian, >>> >>> thanks for the info. I am a step further, but it cannot load the grammar >>> files. >>> I am sending through event_socket: >>> >>> SendMsg >>> call-command: execute >>> execute-app-name: detect_speech >>> execute-app-arg: pocketsphinx yes no >>> >>> However I get the message (also when I am using Pizza demo): >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() >>> sofia/internal/1000 at sip2.server.com Command Execute >>> detect_speech(pocketsphinx yes no) >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 >>> pocketsphinx_asr_load_grammar() Can't open language model >>> /usr/local/freeswitch/grammar/model/communicator. >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 >>> switch_ivr_detect_speech() Error loading Grammar >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 >>> pocketsphinx_asr_close() Port Closed. >>> >>> However the grammar files are there: >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al >>> total 12752 >>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . >>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. >>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING >>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params >>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means >>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict >>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump >>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances >>> >>> >>> Any hint? >>> >>> Best regards >>> Peter >>> >>> Brian West schrieb: >>> >>> >>>> You can accomplish this .... here is an example using ESL in perl >>>> >>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 >>>> >>>> /b >>>> >>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: >>>> >>>> >>>> >>>> >>>>> Or back to the basics: Is it possible to use pocketsphinx through >>>>> event >>>>> socket? >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steveu at coppice.org Mon Mar 2 16:08:27 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 03 Mar 2009 08:08:27 +0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: <49AC74FB.3030903@coppice.org> Andy Spitzer wrote: > Woof! > > On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote: > > >> NO. You want something that people THINK exists and works well... >> Reliable human/voice detection doesn't exist in ANY form. >> > > I beg to differ. See http://www.freepatentsonline.com/5521967.html for one way to do it. It works rather well and can quickly descriminate between voice and tone. I've no idea who owns that patent now (not me, for sure). > Since when did a patent mean a problem is solved? For things like speech recognition you can achieve pretty high accuracy in voice detection, but in that case you can delay the audio and make decisions that span the start of the speech burst. For most telephony purposes you need to make a decision on the very first frame of speech, as you can't afford to add latency. That turns it into a tough problem. Something like the VAD in G.729 is about the best people can currently do, but its far from perfect. > There is a simpler, less reliable way of differentiating voice from tone, that as far as I know isn't patented. If you compare the RMS power levels of sequential 40 mS periods, call progress tones will have very consistent power levels from sample to sample. So if 5 or more 40 mS periods have about the same power measurement (within say, 2%), it's a tone. Voice will have dramatic power level differences over that same period. This works very well in today's telephony environment, where tones are computer generated. In the old days when ringback tone was generated off the audio hum from the 20 Hz ring voltage generator...not so well. > That is *not* VAD. What you describe just says "is its energy steady". I will trigger on music, background noise and maybe even some of the fast pulsed tone signals. A proper VAD won't. Regards, Steve From steveu at coppice.org Mon Mar 2 16:10:50 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 03 Mar 2009 08:10:50 +0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> Message-ID: <49AC758A.60304@coppice.org> Hi, mod_vmd is a bit more sophisticated than that. It looks for the signal being narrowband energy. However, mod_vmd isn't very reliable, as it takes a rather high SNR for its narrowband detector to work. So high that a lossy codec like G.711 can barely manage it. Regards, Steve Anthony Minessale wrote: > i think that's what mod_vmd does > > On Mon, Mar 2, 2009 at 11:16 AM, Andy Spitzer > wrote: > > Woof! > > On Sun, 01 Mar 2009 21:28:18 -0500, Brian West > > wrote: > > > NO. You want something that people THINK exists and works well... > > Reliable human/voice detection doesn't exist in ANY form. > > I beg to differ. See > http://www.freepatentsonline.com/5521967.html for one way to do > it. It works rather well and can quickly descriminate between > voice and tone. I've no idea who owns that patent now (not me, > for sure). > > There is a simpler, less reliable way of differentiating voice > from tone, that as far as I know isn't patented. If you compare > the RMS power levels of sequential 40 mS periods, call progress > tones will have very consistent power levels from sample to > sample. So if 5 or more 40 mS periods have about the same power > measurement (within say, 2%), it's a tone. Voice will have > dramatic power level differences over that same period. This > works very well in today's telephony environment, where tones are > computer generated. In the old days when ringback tone was > generated off the audio hum from the 20 Hz ring voltage > generator...not so well. > > --Woof! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveu at coppice.org Mon Mar 2 16:32:53 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 03 Mar 2009 08:32:53 +0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: <49AC7AB5.5060505@coppice.org> Andy Spitzer wrote: > Woof! > > On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote: > > >> NO. You want something that people THINK exists and works well... >> Reliable human/voice detection doesn't exist in ANY form. >> > > I beg to differ. See http://www.freepatentsonline.com/5521967.html for one way to do it. It works rather well and can quickly descriminate between voice and tone. I've no idea who owns that patent now (not me, for sure). > I just had a look through that patent. Its amazing. There is a lot of focussed descriptive text, but a patent only really consists of its claims. Those claims are astonishingly open-ended, and characterise what people had been doing for many years before it was filed - "Well we, er, make a call, we listen for some beeping, and we may hang up based on that". That is a really sick patent. Steve From mszlazak at aol.com Mon Mar 2 16:49:09 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 02 Mar 2009 19:49:09 -0500 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AC6EE4.9080509@gmx.net> References: <49AC6EE4.9080509@gmx.net> Message-ID: <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> I think you need to talk to Brian. Apparently this is a "new" pocketsphinx which works on a different format from those found in the pizza demo. Also, pocketsphinx crashes if it "hears" anything outside the grammar which apparently is a longstanding bug. Brian mentioned they are working on getting this fixed. I kept getting: 2009-02-25 19:49:32 [ERR] mod_pocketsphinx.c:140 pocketsphinx_asr_load_grammar() Can't open dictionary C:\Source\freeswitch-snapshot\Debug\grammar\default.dic. 2009-02-25 19:49:32 [WARNING] mod_pocketsphinx.c:219 pocketsphinx_asr_close() Port Closed. The suggestion was to "Just copy the cmudict.0.6d to default.dic, not sure how well it will perform on windows.. if it does badly you can slim the dictionary down to words you know you'll be using." https://cmusphinx.svn.sourceforge.net/svnroot/cmusphinx/trunk/cmudict/cmudict.0.6d That gave me more problems so I'm waiting for the fix. Mark. -----Original Message----- From: Peter P GMX To: freeswitch-users at lists.freeswitch.org Sent: Mon, 2 Mar 2009 3:42 pm Subject: Re: [Freeswitch-users] pocketsphinx and event socket Thanks Addison. The Pizza files are there (as mentionned is it a copy of an already working system). In fact freeswitch is complaning about /usr/local/freeswitch/grammar/model/communicator which he cannot load So somehow freeswitch is not willing to open the files, but I have no clue why. So any hints are welcome. Best regards Peter Addison Martin schrieb: > Peter, > > You need the grammar files for the pizza demo: > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo > has lonks to premade fles for everyhting to get the pizza demo working > with pocketshinx. Those to not come with the source code when you > update from SVN. > > Nik > > > > On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX wrote: > >> Some more info: >> the system I am working on is a copy (dd copy) of a system where the >> pizza demo works on. >> The only thing I changed was to update to the current freeswitch trunk >> 12293 (it was 10003 before). >> >> Do I need to update the model? I did a make in the model directory, but >> no change. >> >> Best regards >> Peter >> >> Peter P GMX schrieb: >> >>> Hello Brian, >>> >>> thanks for the info. I am a step further, but it cannot load the grammar >>> files. >>> I am sending through event_socket: >>> >>> SendMsg >>> call-command: execute >>> execute-app-name: detect_speech >>> execute-app-arg: pocketsphinx yes no >>> >>> However I get the message (also when I am using Pizza demo): >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() >>> sofia/internal/1000 at sip2.server.com Command Execute >>> detect_speech(pocketsphinx yes no) >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 >>> pocketsphinx_asr_load_grammar() Can't open language model >>> /usr/local/freeswitch/grammar/model/communicator. >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 >>> switch_ivr_detect_speech() Error loading Grammar >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 >>> pocketsphinx_asr_close() Port Closed. >>> >>> However the grammar files are there: >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al >>> total 12752 >>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . >>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. >>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING >>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params >>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means >>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict >>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump >>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances >>> >>> >>> Any hint? >>> >>> Best regards >>> Peter >>> >>> Brian West schrieb: >>> >>> >>>> You can accomplish this .... here is an example using ESL in perl >>>> >>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 >>>> >>>> /b >>>> >>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: >>>> >>>> >>>> >>>> >>>>> Or back to the basics: Is it possible to use pocketsphinx through >>>>> event >>>>> socket? >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/27f69544/attachment-0001.html From woof at nortel.com Mon Mar 2 16:51:01 2009 From: woof at nortel.com (Andy Spitzer) Date: Mon, 02 Mar 2009 19:51:01 -0500 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <49AC7AB5.5060505@coppice.org> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <49AC7AB5.5060505@coppice.org> Message-ID: Woof! On Mon, 02 Mar 2009 19:32:53 -0500, Steve Underwood wrote: I just had a look through that patent. Its amazing. There is a lot of > focussed descriptive text, but a patent only really consists of its > claims. Those claims are astonishingly open-ended, and characterise what > people had been doing for many years before it was filed - "Well we, er, > make a call, we listen for some beeping, and we may hang up based on > that". That is a really sick patent. Yep, I agree. It was the ferping lawyers who kept "adding value" to try to broaden it. What we (the inventors) wrote up was nice and clean. It does have some new and novel technical approaches that we really did come up with...and could find no prior art for. Then the lawyers got to it. A true example of what's wrong with software patents these days. --Woof! From kawarod at laposte.net Mon Mar 2 23:07:32 2009 From: kawarod at laposte.net (rod) Date: Tue, 03 Mar 2009 11:07:32 +0400 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: References: <49AB8562.4050806@laposte.net> Message-ID: <49ACD734.7000700@laposte.net> Hi Michael, I checked on wiki, is the following the good way to go (sorry I'm not very familiar with your debugging tool). $ gdb bin/freeswitch core.xxx bt bt full thread apply all bt thread apply all bt full If I understand well I have to rerun the tests, as I did not start FS using GDB. regards, rod Michael Jerris wrote: > Could you please post this to jira along with a thread apply all bt of > a core file taken from the process with the stuck sessions. > > Mike > > On Mar 2, 2009, at 2:06 AM, rod wrote: > > >> Hi All, >> >> I ran some longer tests with FS 1.0.3 acting as an SBC. >> The test machine has the following specs: >> - Intel Quad Core Q9550 >> - 8GB RAM (far too much from what I saw) >> >> After 3 days running SIPP with 750 simultaneous calls (1500 >> channels) at >> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. >> >> In the CLI, using status command I got this: >> >> freeswitch at internal> status >> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds, >> 607 microseconds >> 15817560 session(s) since startup >> 22 session(s) 0/500 >> >> But when I use "show channels" or "show calls", I see nothing. So I'm >> wondering where are these 22 sessions ? >> >> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs. >> >> Successful call --> 5271434 >> Failed call ---> 1554 (less than 0.03%) >> >> regards, >> rod. >> >> >> >> complete SIPP summary: >> >> ------------------------------ Scenario Screen -------- [1-9]: Change >> Screen -- >> Call-rate(length) Port Total-time Total-calls Remote-host >> 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 >> 10.10.10.254:5060(UDP) >> >> 0 new calls during 0.856 s period 7 ms scheduler resolution >> 0 calls (limit 750) Peak was 750 calls, after 15 s >> 0 Running, 34 Paused, 0 Woken up >> 15544 out-of-call msg (discarded) >> 1 open sockets >> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate >> (kB/s) >> 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate >> (kB/s) >> >> Messages Retrans Timeout >> Unexpected-Msg >> INVITE ----------> 5273022 0 0 >> 100 <---------- 5273022 0 1554 >> 180 <---------- 0 0 0 >> 183 <---------- 0 0 0 >> 200 <---------- E-RTD1 5271434 0 0 >> ACK ----------> 5271434 0 >> Pause [ 35.0s] 5271434 0 >> BYE ----------> 5271434 0 0 >> 200 <---------- 5271434 0 0 >> >> ------------------------------ Test Terminated >> -------------------------------- >> >> >> ----------------------------- Statistics Screen ------- [1-9]: Change >> Screen -- >> Start Time | 2009-02-27 >> 09:11:31 >> Last Reset Time | 2009-03-02 >> 07:49:10 >> Current Time | 2009-03-02 >> 07:49:11 >> -------------------------+--------------------------- >> +-------------------------- >> Counter Name | Periodic value | Cumulative value >> -------------------------+--------------------------- >> +-------------------------- >> Elapsed Time | 00:00:00:857 | >> 70:37:39:429 >> Call Rate | 0.000 cps | 20.739 >> cps >> -------------------------+--------------------------- >> +-------------------------- >> Incoming call created | 0 | >> 0 >> OutGoing call created | 0 | >> 5273022 >> Total Call created | | >> 5273022 >> Current Call | 34 >> | >> -------------------------+--------------------------- >> +-------------------------- >> Successful call | 0 | >> 5271434 >> Failed call | 0 | >> 1554 >> -------------------------+--------------------------- >> +-------------------------- >> Response Time 1 | 00:00:00:000 | >> 00:00:00:240 >> Call Length | 38:32:13:386 | >> 00:00:36:131 >> ------------------------------ Test Terminated >> -------------------------------- >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From mrene_lists at avgs.ca Mon Mar 2 23:56:01 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 3 Mar 2009 02:56:01 -0500 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: <49ACD734.7000700@laposte.net> References: <49AB8562.4050806@laposte.net> <49ACD734.7000700@laposte.net> Message-ID: <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca> Yes, you may also link (or copy) the .gdbinit file found in the support-d folder to your home directory. This is going to enable some GDB macros written for FS. Once thats done you can do the following commands and include them too: list_sessions hash_it_str_x session_manager.session_table switch_core_session_t channel->state Its important to know that what you see in "show channels" and "show calls" is just a DB query to sqlite, Those commands will go directly in the core and list those sessions. Math On 3-Mar-09, at 2:07 AM, rod wrote: > Hi Michael, > > I checked on wiki, is the following the good way to go (sorry I'm not > very familiar with your debugging tool). > > $ gdb bin/freeswitch core.xxx > > bt > bt full > thread apply all bt > thread apply all bt full > > > If I understand well I have to rerun the tests, as I did not start FS > using GDB. > > regards, > rod > > > > > Michael Jerris wrote: >> Could you please post this to jira along with a thread apply all bt >> of >> a core file taken from the process with the stuck sessions. >> >> Mike >> >> On Mar 2, 2009, at 2:06 AM, rod wrote: >> >> >>> Hi All, >>> >>> I ran some longer tests with FS 1.0.3 acting as an SBC. >>> The test machine has the following specs: >>> - Intel Quad Core Q9550 >>> - 8GB RAM (far too much from what I saw) >>> >>> After 3 days running SIPP with 750 simultaneous calls (1500 >>> channels) at >>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. >>> >>> In the CLI, using status command I got this: >>> >>> freeswitch at internal> status >>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 >>> milliseconds, >>> 607 microseconds >>> 15817560 session(s) since startup >>> 22 session(s) 0/500 >>> >>> But when I use "show channels" or "show calls", I see nothing. So >>> I'm >>> wondering where are these 22 sessions ? >>> >>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free >>> CPUs. >>> >>> Successful call --> 5271434 >>> Failed call ---> 1554 (less than 0.03%) >>> >>> regards, >>> rod. >>> >>> >>> >>> complete SIPP summary: >>> >>> ------------------------------ Scenario Screen -------- [1-9]: >>> Change >>> Screen -- >>> Call-rate(length) Port Total-time Total-calls Remote-host >>> 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 >>> 10.10.10.254:5060(UDP) >>> >>> 0 new calls during 0.856 s period 7 ms scheduler resolution >>> 0 calls (limit 750) Peak was 750 calls, after >>> 15 s >>> 0 Running, 34 Paused, 0 Woken up >>> 15544 out-of-call msg (discarded) >>> 1 open sockets >>> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP >>> rate >>> (kB/s) >>> 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate >>> (kB/s) >>> >>> Messages Retrans Timeout >>> Unexpected-Msg >>> INVITE ----------> 5273022 0 0 >>> 100 <---------- 5273022 0 1554 >>> 180 <---------- 0 0 0 >>> 183 <---------- 0 0 0 >>> 200 <---------- E-RTD1 5271434 0 0 >>> ACK ----------> 5271434 0 >>> Pause [ 35.0s] 5271434 0 >>> BYE ----------> 5271434 0 0 >>> 200 <---------- 5271434 0 0 >>> >>> ------------------------------ Test Terminated >>> -------------------------------- >>> >>> >>> ----------------------------- Statistics Screen ------- [1-9]: >>> Change >>> Screen -- >>> Start Time | 2009-02-27 >>> 09:11:31 >>> Last Reset Time | 2009-03-02 >>> 07:49:10 >>> Current Time | 2009-03-02 >>> 07:49:11 >>> -------------------------+--------------------------- >>> +-------------------------- >>> Counter Name | Periodic value | Cumulative >>> value >>> -------------------------+--------------------------- >>> +-------------------------- >>> Elapsed Time | 00:00:00:857 | >>> 70:37:39:429 >>> Call Rate | 0.000 cps | 20.739 >>> cps >>> -------------------------+--------------------------- >>> +-------------------------- >>> Incoming call created | 0 | >>> 0 >>> OutGoing call created | 0 | >>> 5273022 >>> Total Call created | | >>> 5273022 >>> Current Call | 34 >>> | >>> -------------------------+--------------------------- >>> +-------------------------- >>> Successful call | 0 | >>> 5271434 >>> Failed call | 0 | >>> 1554 >>> -------------------------+--------------------------- >>> +-------------------------- >>> Response Time 1 | 00:00:00:000 | >>> 00:00:00:240 >>> Call Length | 38:32:13:386 | >>> 00:00:36:131 >>> ------------------------------ Test Terminated >>> -------------------------------- >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Tue Mar 3 00:16:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 03 Mar 2009 09:16:02 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> Message-ID: <49ACE742.5090809@ewetel.de> Hi Mike, no, I want just one file in stereo, where left channel is caller and right channel is callee. regards Helmut On 02.03.2009 19:05, Michael Collins wrote: > > Just to confirm - you are trying to record each leg of the call into a > separate file? In other words, one call creates two separate audio > recordings? > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090303/d8383e93/attachment.html From Prometheus001 at gmx.net Tue Mar 3 03:07:39 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 03 Mar 2009 12:07:39 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> Message-ID: <49AD0F7B.7000802@gmx.net> Thanks Mark, I now switched back to rev. 10003 and the Pizza works again. Best regards Petere mszlazak at aol.com schrieb: > I think you need to talk to Brian. > > Apparently this is a "new" pocketsphinx which works on a different > format from those found in the pizza demo. > > Also, pocketsphinx crashes if it "hears" anything outside the grammar > which apparently is a longstanding bug. Brian mentioned they are > working on getting this fixed. > > I kept getting: > > 2009-02-25 19:49:32 [ERR] mod_pocketsphinx.c:140 > pocketsphinx_asr_load_grammar() Can't open dictionary > C:\Source\freeswitch-snapshot\Debug\grammar\default.dic. > 2009-02-25 19:49:32 [WARNING] mod_pocketsphinx.c:219 > pocketsphinx_asr_close() Port Closed. > > The suggestion was to "Just copy the cmudict.0.6d to default.dic, not > sure how well it will perform on windows.. if it does badly you can > slim the dictionary down to words you know you'll be using." > > https://cmusphinx.svn.sourceforge.net/svnroot/cmusphinx/trunk/cmudict/cmudict.0.6d > > That gave me more problems so I'm waiting for the fix. > > Mark. > > > > -----Original Message----- > From: Peter P GMX > To: freeswitch-users at lists.freeswitch.org > Sent: Mon, 2 Mar 2009 3:42 pm > Subject: Re: [Freeswitch-users] pocketsphinx and event socket > > Thanks Addison. > > > > > > > > The Pizza files are there (as mentionned is it a copy of an already > > > > > > > > working system). > > > > > > > > In fact freeswitch is complaning about > > > > > > > > /usr/local/freeswitch/grammar/model/communicator which he cannot load > > > > > > > > > > > > > > > > So somehow freeswitch is not willing to open the files, but I have no > > > > > > > > clue why. So any hints are welcome. > > > > > > > > > > > > > > > > Best regards > > > > > > > > Peter > > > > > > > > > > > > > > > > > > > > > > > > Addison Martin schrieb: > > > > > > > > > Peter, > > > > > > > > > > > > > > > > > > You need the grammar files for the pizza demo: > > > > > > > > > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo > > > > > > > > > has lonks to premade fles for everyhting to get the pizza demo working > > > > > > > > > with pocketshinx. Those to not come with the source code when you > > > > > > > > > update from SVN. > > > > > > > > > > > > > > > > > > Nik > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX > wrote: > > > > > > > > > > > > > > > > > >> Some more info: > > > > > > > > >> the system I am working on is a copy (dd copy) of a system where the > > > > > > > > >> pizza demo works on. > > > > > > > > >> The only thing I changed was to update to the current freeswitch trunk > > > > > > > > >> 12293 (it was 10003 before). > > > > > > > > >> > > > > > > > > >> Do I need to update the model? I did a make in the model directory, but > > > > > > > > >> no change. > > > > > > > > >> > > > > > > > > >> Best regards > > > > > > > > >> Peter > > > > > > > > >> > > > > > > > > >> Peter P GMX schrieb: > > > > > > > > >> > > > > > > > > >>> Hello Brian, > > > > > > > > >>> > > > > > > > > >>> thanks for the info. I am a step further, but it cannot load the grammar > > > > > > > > >>> files. > > > > > > > > >>> I am sending through event_socket: > > > > > > > > >>> > > > > > > > > >>> SendMsg > > > > > > > > >>> call-command: execute > > > > > > > > >>> execute-app-name: detect_speech > > > > > > > > >>> execute-app-arg: pocketsphinx yes no > > > > > > > > >>> > > > > > > > > >>> However I get the message (also when I am using Pizza demo): > > > > > > > > >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() > > > > > > > > >>> sofia/internal/1000 at sip2.server.com Command Execute > > > > > > > > >>> detect_speech(pocketsphinx yes no) > > > > > > > > >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 > > > > > > > > >>> pocketsphinx_asr_load_grammar() Can't open language model > > > > > > > > >>> /usr/local/freeswitch/grammar/model/communicator. > > > > > > > > >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 > > > > > > > > >>> switch_ivr_detect_speech() Error loading Grammar > > > > > > > > >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 > > > > > > > > >>> pocketsphinx_asr_close() Port Closed. > > > > > > > > >>> > > > > > > > > >>> However the grammar files are there: > > > > > > > > >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# > > > > > > > > >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al > > > > > > > > >>> total 12752 > > > > > > > > >>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . > > > > > > > > >>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances > > > > > > > > >>> > > > > > > > > >>> > > > > > > > > >>> Any hint? > > > > > > > > >>> > > > > > > > > >>> Best regards > > > > > > > > >>> Peter > > > > > > > > >>> > > > > > > > > >>> Brian West schrieb: > > > > > > > > >>> > > > > > > > > >>> > > > > > > > > >>>> You can accomplish this .... here is an example using ESL in perl > > > > > > > > >>>> > > > > > > > > >>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 > > > > > > > > >>>> > > > > > > > > >>>> /b > > > > > > > > >>>> > > > > > > > > >>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: > > > > > > > > >>>> > > > > > > > > >>>> > > > > > > > > >>>> > > > > > > > > >>>> > > > > > > > > >>>>> Or back to the basics: Is it possible to use pocketsphinx through > > > > > > > > >>>>> event > > > > > > > > >>>>> socket? > > > > > > > > >>>>> > > > > > > > > >>>>> > > > > > > > > >>>>> > > > > > > > > >>>> _______________________________________________ > > > > > > > > >>>> Freeswitch-users mailing list > > > > > > > > >>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > http://www.freeswitch.org > > > > > > > > > ------------------------------------------------------------------------ > *A Good Credit Score is 700 or Above. See yours in just 2 easy steps! > * > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Mar 3 06:22:42 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 08:22:42 -0600 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49ACE742.5090809@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> Message-ID: That is exactly what it does. I have confirmed it works can you please file a jira with examples and info of what your experiencing? /b On Mar 3, 2009, at 2:16 AM, Helmut Kuper wrote: > Hi Mike, > > no, I want just one file in stereo, where left channel is caller and > right channel is callee. > > regards > Helmut From brian at freeswitch.org Tue Mar 3 06:23:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 08:23:26 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AD0F7B.7000802@gmx.net> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> Message-ID: <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> It works if you have the latest SVN with the new grammar files in jsgf format. http://www.bkw.org/pizza_gram.tar.gz /b On Mar 3, 2009, at 5:07 AM, Peter P GMX wrote: > Thanks Mark, > > I now switched back to rev. 10003 and the Pizza works again. > > Best regards > Petere From brian at freeswitch.org Tue Mar 3 06:24:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 08:24:07 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AC6EE4.9080509@gmx.net> References: <49A92BAE.4090907@gmx.net> <49AB0BCD.8030108@gmx.net> <49AC4233.6060506@gmx.net> <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com> <49AC6EE4.9080509@gmx.net> Message-ID: Looks like the acoustical model wasn't installed... you might need to remove all references of pocketsphinx and sphinxbase from libs and let it redownload them all. /b On Mar 2, 2009, at 5:42 PM, Peter P GMX wrote: > Thanks Addison. > The Pizza files are there (as mentionned is it a copy of an already > working system). > In fact freeswitch is complaning about > /usr/local/freeswitch/grammar/model/communicator which he cannot load > > So somehow freeswitch is not willing to open the files, but I have no > clue why. So any hints are welcome. > > Best regards > Peter From helmut.kuper at ewetel.de Tue Mar 3 07:24:26 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 03 Mar 2009 16:24:26 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> Message-ID: <49AD4BAA.8090208@ewetel.de> Hello Brian, you are right. It works, had to use a different player. But caller is also there much louder than callee. Any way to tune this? regards helmut On 03.03.2009 15:22, Brian West wrote: > That is exactly what it does. I have confirmed it works can you > please file a jira with examples and info of what your experiencing? From brian at freeswitch.org Tue Mar 3 07:40:25 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 09:40:25 -0600 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49AD4BAA.8090208@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> Message-ID: <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> Thats going to depend on a lot of issues... are you on SVN trunk? What codecs? What is the path? /b On Mar 3, 2009, at 9:24 AM, Helmut Kuper wrote: > Hello Brian, > > you are right. It works, had to use a different player. But caller is > also there much louder than callee. Any way to tune this? > > regards > helmut From kerrada2003 at yahoo.com Tue Mar 3 07:53:45 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Tue, 3 Mar 2009 07:53:45 -0800 (PST) Subject: [Freeswitch-users] Mandatory IE missing In-Reply-To: Message-ID: <590483.92469.qm@web33703.mail.mud.yahoo.com> Hi, During proxy authentication, I got "mandatory IE missing" error in the response as shown below. How this error can be resolved? recv 585 bytes from udp/[209.82.10.250]:1898 at 15:31:39.933782: ?? ------------------------------------------------------------------------ ?? INVITE sip:9999 at 209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: sip:9999 at 209.82.10.235 ?? Contact: sip:1002 at 209.82.10.250:1091 ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988828 INVITE ?? Content-Length: 187 ?? Content-Type: application/sdp ?? User-Agent: HelpCaster SoftPHONE ?? Supported: com.hearme.mux ? ?? v=0 ?? o=HelpCaster 153553387 153553387 IN IP4 192.168.10.31 ?? s=HelpCaster ?? c=IN IP4 209.82.10.250 ?? t=0 0 ?? m=audio 8002 RTP/AVP 0 4 101 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-15 ?? ------------------------------------------------------------------------ send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.934100: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: sip:9999 at 209.82.10.235 ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988828 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ send 746 bytes to udp/[209.82.10.250]:1091 at 15:31:39.938913: ?? ------------------------------------------------------------------------ ?? SIP/2.0 407 Proxy Authentication Required ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: ;tag=K83mZmZH8g3Hr ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988828 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Accept: application/sdp ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer ?? Proxy-Authenticate: Digest realm="209.82.10.235", nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54" ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.949517: ?? ------------------------------------------------------------------------ ?? ACK sip:9999 at 209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: ;tag=K83mZmZH8g3Hr ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988828 ACK ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 584 bytes from udp/[209.82.10.250]:1898 at 15:31:39.953137: ?? ------------------------------------------------------------------------ ?? INVITE sip:9999 at 209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: sip:9999 at 209.82.10.235 ?? Contact: sip:1002 at 209.82.10.250:1091 ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988829 INVITE ?? Content-Length: 187 ?? Content-Type: application/sdp ?? User-Agent: HelpCaster SoftPHONE ?? Supported: com.hearme.mux ?? Proxy-Authorization:? Digest username="1002",realm="209.82.10.235",nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54",response=" d1614cd024acab0b794751285f5cb1fe",uri="sip:9999 at 209.82.10.235" ? ?? ------------------------------------------------------------------------ send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.953336: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: sip:9999 at 209.82.10.235 ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988829 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ send 696 bytes to udp/[209.82.10.250]:1091 at 15:31:39.956016: ?? ------------------------------------------------------------------------ ?? SIP/2.0 480 Temporarily Unavailable ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: ;tag=mHXD1FgN5SS4K ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988829 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Accept: application/sdp ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer ?? Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING" ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.964632: ?? ------------------------------------------------------------------------ ?? ACK sip:9999 at 209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: ;tag=mHXD1FgN5SS4K ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988829 ACK ?? Content-Length: 0 Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090303/39844287/attachment-0001.html From freeswitch at servercorps.com Tue Mar 3 08:42:01 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Tue, 3 Mar 2009 10:42:01 -0600 Subject: [Freeswitch-users] First time test set up FreeSwitch and SPA3102/SPA3000 In-Reply-To: <9ed22e920903012343u5e50c89fkfa9d1127894199c@mail.gmail.com> References: <9ed22e920903012343u5e50c89fkfa9d1127894199c@mail.gmail.com> Message-ID: <92e7d2090903030842p509573co969f89a3fb72a8ad@mail.gmail.com> api, It may help if we knew a little about what you are trying to do. Could you explain a bit why you think you need a PBX? What re your goals, and objective? Knowing that may help us get you the information you need. Regards, Nik On Mon, Mar 2, 2009 at 1:43 AM, Aplayful Idiot wrote: > I have no background in telephony but probably need to use a PBX. > > FreeSwitch was recommended by a casual contact so I would like to start > first by setting up a small test. > > I have a SPA3102 attached to the box running FS and to a ordinary phone > line. I registered SPA in conf/directory/default/line1.xml and it works ok > but I can't get caller id numbers from incoming calls. All FS sees is > "line1" which is found in file line1.xml as name="effective_caller_id_number" value="line1"/>. > > Looking back over the FS wiki, I'm now wondering if the SPA should of been > set-up as a gateway but reading it is confusing at least to me. Sometimes I > think the analogue-line-SPA-FS is like a softphone which is registered to an > extension numbered xml file in conf/directory/default/ but then issues like > not getting outside incoming caller id's makes me think I've got this all > wrong. > > Can someone help me out with this? > > api > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Tue Mar 3 13:17:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Mar 2009 15:17:30 -0600 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca> References: <49AB8562.4050806@laposte.net> <49ACD734.7000700@laposte.net> <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca> Message-ID: <191c3a030903031317y55ce2c9bsb4a93518db261973@mail.gmail.com> and you may want to update and try trunk to make sure it's not fixed On Tue, Mar 3, 2009 at 1:56 AM, Mathieu Rene wrote: > Yes, you may also link (or copy) the .gdbinit file found in the > support-d folder to your home directory. > This is going to enable some GDB macros written for FS. > > Once thats done you can do the following commands and include them too: > > list_sessions > > hash_it_str_x session_manager.session_table switch_core_session_t > channel->state > > > Its important to know that what you see in "show channels" and "show > calls" is just a DB query to sqlite, Those commands will go directly > in the core and list those sessions. > > Math > > On 3-Mar-09, at 2:07 AM, rod wrote: > > > Hi Michael, > > > > I checked on wiki, is the following the good way to go (sorry I'm not > > very familiar with your debugging tool). > > > > $ gdb bin/freeswitch core.xxx > > > > bt > > bt full > > thread apply all bt > > thread apply all bt full > > > > > > If I understand well I have to rerun the tests, as I did not start FS > > using GDB. > > > > regards, > > rod > > > > > > > > > > Michael Jerris wrote: > >> Could you please post this to jira along with a thread apply all bt > >> of > >> a core file taken from the process with the stuck sessions. > >> > >> Mike > >> > >> On Mar 2, 2009, at 2:06 AM, rod wrote: > >> > >> > >>> Hi All, > >>> > >>> I ran some longer tests with FS 1.0.3 acting as an SBC. > >>> The test machine has the following specs: > >>> - Intel Quad Core Q9550 > >>> - 8GB RAM (far too much from what I saw) > >>> > >>> After 3 days running SIPP with 750 simultaneous calls (1500 > >>> channels) at > >>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. > >>> > >>> In the CLI, using status command I got this: > >>> > >>> freeswitch at internal> status > >>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 > >>> milliseconds, > >>> 607 microseconds > >>> 15817560 session(s) since startup > >>> 22 session(s) 0/500 > >>> > >>> But when I use "show channels" or "show calls", I see nothing. So > >>> I'm > >>> wondering where are these 22 sessions ? > >>> > >>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free > >>> CPUs. > >>> > >>> Successful call --> 5271434 > >>> Failed call ---> 1554 (less than 0.03%) > >>> > >>> regards, > >>> rod. > >>> > >>> > >>> > >>> complete SIPP summary: > >>> > >>> ------------------------------ Scenario Screen -------- [1-9]: > >>> Change > >>> Screen -- > >>> Call-rate(length) Port Total-time Total-calls Remote-host > >>> 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 > >>> 10.10.10.254:5060(UDP) > >>> > >>> 0 new calls during 0.856 s period 7 ms scheduler resolution > >>> 0 calls (limit 750) Peak was 750 calls, after > >>> 15 s > >>> 0 Running, 34 Paused, 0 Woken up > >>> 15544 out-of-call msg (discarded) > >>> 1 open sockets > >>> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP > >>> rate > >>> (kB/s) > >>> 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate > >>> (kB/s) > >>> > >>> Messages Retrans Timeout > >>> Unexpected-Msg > >>> INVITE ----------> 5273022 0 0 > >>> 100 <---------- 5273022 0 1554 > >>> 180 <---------- 0 0 0 > >>> 183 <---------- 0 0 0 > >>> 200 <---------- E-RTD1 5271434 0 0 > >>> ACK ----------> 5271434 0 > >>> Pause [ 35.0s] 5271434 0 > >>> BYE ----------> 5271434 0 0 > >>> 200 <---------- 5271434 0 0 > >>> > >>> ------------------------------ Test Terminated > >>> -------------------------------- > >>> > >>> > >>> ----------------------------- Statistics Screen ------- [1-9]: > >>> Change > >>> Screen -- > >>> Start Time | 2009-02-27 > >>> 09:11:31 > >>> Last Reset Time | 2009-03-02 > >>> 07:49:10 > >>> Current Time | 2009-03-02 > >>> 07:49:11 > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Counter Name | Periodic value | Cumulative > >>> value > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Elapsed Time | 00:00:00:857 | > >>> 70:37:39:429 > >>> Call Rate | 0.000 cps | 20.739 > >>> cps > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Incoming call created | 0 | > >>> 0 > >>> OutGoing call created | 0 | > >>> 5273022 > >>> Total Call created | | > >>> 5273022 > >>> Current Call | 34 > >>> | > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Successful call | 0 | > >>> 5271434 > >>> Failed call | 0 | > >>> 1554 > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Response Time 1 | 00:00:00:000 | > >>> 00:00:00:240 > >>> Call Length | 38:32:13:386 | > >>> 00:00:36:131 > >>> ------------------------------ Test Terminated > >>> -------------------------------- > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090303/bf5f72d7/attachment.html From mike at jerris.com Tue Mar 3 14:08:11 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 3 Mar 2009 17:08:11 -0500 Subject: [Freeswitch-users] Mandatory IE missing In-Reply-To: <590483.92469.qm@web33703.mail.mud.yahoo.com> References: <590483.92469.qm@web33703.mail.mud.yahoo.com> Message-ID: The debug logs should give you more information about what is happening here. Mike On Mar 3, 2009, at 10:53 AM, Ali Al-Rubaie wrote: > Hi, > > During proxy authentication, I got "mandatory IE missing" error in > the response as shown below. How this error can be resolved? From Prometheus001 at gmx.net Tue Mar 3 17:05:40 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 04 Mar 2009 02:05:40 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> Message-ID: <49ADD3E4.20408@gmx.net> Thank you Brian, I will try this later. Currently I was happy to get this working on SVN 10003. As mod_pockesphinx has changed/evolved significantely: Will there also be major changes in the events I receive through mod_eventsocket? I spend some time on parsing the right data out of the eventsocket interface, and I would just have an idea, if I will have to expect significant work to do, when I later switch to the current SVN. Will I need updated grammar files for the other models too? Best regards Peter Brian West schrieb: > It works if you have the latest SVN with the new grammar files in jsgf > format. http://www.bkw.org/pizza_gram.tar.gz > > > /b > > On Mar 3, 2009, at 5:07 AM, Peter P GMX wrote: > > >> Thanks Mark, >> >> I now switched back to rev. 10003 and the Pizza works again. >> >> Best regards >> Petere >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Tue Mar 3 19:00:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 21:00:35 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49ADD3E4.20408@gmx.net> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> <49ADD3E4.20408@gmx.net> Message-ID: <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> Well you should use ESL then ;) /b On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote: > Thank you Brian, > > I will try this later. > > Currently I was happy to get this working on SVN 10003. > > As mod_pockesphinx has changed/evolved significantely: Will there also > be major changes in the events I receive through mod_eventsocket? > I spend some time on parsing the right data out of the eventsocket > interface, and I would just have an idea, if I will have to expect > significant work to do, when I later switch to the current SVN. > > Will I need updated grammar files for the other models too? > > Best regards > Peter From jgarland at jasongarland.com Tue Mar 3 20:12:45 2009 From: jgarland at jasongarland.com (Jason Garland) Date: Tue, 3 Mar 2009 23:12:45 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Qt portaudio interface In-Reply-To: <3A65F8B6-96B4-45EE-80E9-A30AFEED0BB3@avgs.ca> References: <3A65F8B6-96B4-45EE-80E9-A30AFEED0BB3@avgs.ca> Message-ID: I still think a web based interface would work well and be more cross platform. The web app could be served from within FS. http://127.0.0.1:8080/myfancysoftphone/ Sent from my iPhone On Mar 2, 2009, at 12:00 AM, Mathieu Rene wrote: > Hi all, > > Anyone interested in contributing to a Qt interface in order to make a > decent softphone using FS please reply to this thread. > (also give your availability so we can have a conference call to > decide stuff) > > Thanks, > Math > > _______________________________________________ > 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 jgarland at jasongarland.com Tue Mar 3 20:18:40 2009 From: jgarland at jasongarland.com (Jason Garland) Date: Tue, 3 Mar 2009 23:18:40 -0500 Subject: [Freeswitch-users] Mandatory IE missing In-Reply-To: <590483.92469.qm@web33703.mail.mud.yahoo.com> References: <590483.92469.qm@web33703.mail.mud.yahoo.com> Message-ID: <0744A0FC-7FCE-4662-944D-9FE133142AE1@jasongarland.com> Your HelpCaster softphone didn't send any SDP on the second INVITE. Fix the softphone or try turning on late codec negotiation in you Sofia profile. Sent from my iPhone On Mar 3, 2009, at 10:53 AM, Ali Al-Rubaie wrote: > > Hi, > > During proxy authentication, I got "mandatory IE missing" error in > the response as shown below. How this error can be resolved? > > recv 585 bytes from udp/[209.82.10.250]:1898 at 15:31:39.933782: > > > --- > --------------------------------------------------------------------- > > INVITE sip:9999 at 209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: sip:9999 at 209.82.10.235 > > Contact: sip:1002 at 209.82.10.250:1091 > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988828 INVITE > > Content-Length: 187 > > Content-Type: application/sdp > > User-Agent: HelpCaster SoftPHONE > > Supported: com.hearme.mux > > > > v=0 > > o=HelpCaster 153553387 153553387 IN IP4 192.168.10.31 > > s=HelpCaster > > c=IN IP4 209.82.10.250 > > t=0 0 > > m=audio 8002 RTP/AVP 0 4 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > --- > --------------------------------------------------------------------- > > send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.934100: > > > --- > --------------------------------------------------------------------- > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: sip:9999 at 209.82.10.235 > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988828 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > send 746 bytes to udp/[209.82.10.250]:1091 at 15:31:39.938913: > > > --- > --------------------------------------------------------------------- > > SIP/2.0 407 Proxy Authentication Required > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: ;tag=K83mZmZH8g3Hr > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988828 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > > Proxy-Authenticate: Digest realm="209.82.10.235", > nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54" > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.949517: > > > --- > --------------------------------------------------------------------- > > ACK sip:9999 at 209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: ;tag=K83mZmZH8g3Hr > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988828 ACK > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > recv 584 bytes from udp/[209.82.10.250]:1898 at 15:31:39.953137: > > > --- > --------------------------------------------------------------------- > > INVITE sip:9999 at 209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: sip:9999 at 209.82.10.235 > > Contact: sip:1002 at 209.82.10.250:1091 > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988829 INVITE > > Content-Length: 187 > > Content-Type: application/sdp > > User-Agent: HelpCaster SoftPHONE > > Supported: com.hearme.mux > > Proxy-Authorization: Digest > username= > "1002" > ,realm= > "209.82.10.235" > ,nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54",response=" > > d1614cd024acab0b794751285f5cb1fe",uri="sip:9999 at 209.82.10.235" > > > > > --- > --------------------------------------------------------------------- > > send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.953336: > > > --- > --------------------------------------------------------------------- > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: sip:9999 at 209.82.10.235 > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988829 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > send 696 bytes to udp/[209.82.10.250]:1091 at 15:31:39.956016: > > > --- > --------------------------------------------------------------------- > > SIP/2.0 480 Temporarily Unavailable > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: ;tag=mHXD1FgN5SS4K > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988829 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > > Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING" > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.964632: > > > --- > --------------------------------------------------------------------- > > ACK sip:9999 at 209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: ;tag=mHXD1FgN5SS4K > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988829 ACK > > Content-Length: 0 > > > Thanks! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090303/3d1cc9e8/attachment.html From brian at freeswitch.org Tue Mar 3 20:21:08 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 22:21:08 -0600 Subject: [Freeswitch-users] Mandatory IE missing In-Reply-To: <0744A0FC-7FCE-4662-944D-9FE133142AE1@jasongarland.com> References: <590483.92469.qm@web33703.mail.mud.yahoo.com> <0744A0FC-7FCE-4662-944D-9FE133142AE1@jasongarland.com> Message-ID: <3E391AC2-BF15-4987-B143-5CF05855693E@freeswitch.org> Jason thanks.. you know I think we all missed that one. /b On Mar 3, 2009, at 10:18 PM, Jason Garland wrote: > Your HelpCaster softphone didn't send any SDP on the second INVITE. > Fix the softphone or try turning on late codec negotiation in you > Sofia profile. > > Sent from my iPhone From kawarod at laposte.net Tue Mar 3 22:47:21 2009 From: kawarod at laposte.net (rod) Date: Wed, 04 Mar 2009 10:47:21 +0400 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: <191c3a030903031317y55ce2c9bsb4a93518db261973@mail.gmail.com> References: <49AB8562.4050806@laposte.net> <49ACD734.7000700@laposte.net> <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca> <191c3a030903031317y55ce2c9bsb4a93518db261973@mail.gmail.com> Message-ID: <49AE23F9.7050108@laposte.net> Hi, I'm already trying this. I will update this thread next week. regards. Anthony Minessale wrote: > and you may want to update and try trunk to make sure it's not fixed > > On Tue, Mar 3, 2009 at 1:56 AM, Mathieu Rene > wrote: > > Yes, you may also link (or copy) the .gdbinit file found in the > support-d folder to your home directory. > This is going to enable some GDB macros written for FS. > > Once thats done you can do the following commands and include them > too: > > list_sessions > > hash_it_str_x session_manager.session_table switch_core_session_t > channel->state > > > Its important to know that what you see in "show channels" and "show > calls" is just a DB query to sqlite, Those commands will go directly > in the core and list those sessions. > > Math > > On 3-Mar-09, at 2:07 AM, rod wrote: > > > Hi Michael, > > > > I checked on wiki, is the following the good way to go (sorry > I'm not > > very familiar with your debugging tool). > > > > $ gdb bin/freeswitch core.xxx > > > > bt > > bt full > > thread apply all bt > > thread apply all bt full > > > > > > If I understand well I have to rerun the tests, as I did not > start FS > > using GDB. > > > > regards, > > rod > > > > > > > > > > Michael Jerris wrote: > >> Could you please post this to jira along with a thread apply all bt > >> of > >> a core file taken from the process with the stuck sessions. > >> > >> Mike > >> > >> On Mar 2, 2009, at 2:06 AM, rod wrote: > >> > >> > >>> Hi All, > >>> > >>> I ran some longer tests with FS 1.0.3 acting as an SBC. > >>> The test machine has the following specs: > >>> - Intel Quad Core Q9550 > >>> - 8GB RAM (far too much from what I saw) > >>> > >>> After 3 days running SIPP with 750 simultaneous calls (1500 > >>> channels) at > >>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. > >>> > >>> In the CLI, using status command I got this: > >>> > >>> freeswitch at internal> status > >>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 > >>> milliseconds, > >>> 607 microseconds > >>> 15817560 session(s) since startup > >>> 22 session(s) 0/500 > >>> > >>> But when I use "show channels" or "show calls", I see nothing. So > >>> I'm > >>> wondering where are these 22 sessions ? > >>> > >>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free > >>> CPUs. > >>> > >>> Successful call --> 5271434 > >>> Failed call ---> 1554 (less than 0.03%) > >>> > >>> regards, > >>> rod. > >>> > >>> > >>> > >>> complete SIPP summary: > >>> > >>> ------------------------------ Scenario Screen -------- [1-9]: > >>> Change > >>> Screen -- > >>> Call-rate(length) Port Total-time Total-calls Remote-host > >>> 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 > >>> 10.10.10.254:5060(UDP) > >>> > >>> 0 new calls during 0.856 s period 7 ms scheduler resolution > >>> 0 calls (limit 750) Peak was 750 calls, after > >>> 15 s > >>> 0 Running, 34 Paused, 0 Woken up > >>> 15544 out-of-call msg (discarded) > >>> 1 open sockets > >>> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP > >>> rate > >>> (kB/s) > >>> 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate > >>> (kB/s) > >>> > >>> Messages Retrans Timeout > >>> Unexpected-Msg > >>> INVITE ----------> 5273022 0 0 > >>> 100 <---------- 5273022 0 1554 > >>> 180 <---------- 0 0 0 > >>> 183 <---------- 0 0 0 > >>> 200 <---------- E-RTD1 5271434 0 0 > >>> ACK ----------> 5271434 0 > >>> Pause [ 35.0s] 5271434 0 > >>> BYE ----------> 5271434 0 0 > >>> 200 <---------- 5271434 0 0 > >>> > >>> ------------------------------ Test Terminated > >>> -------------------------------- > >>> > >>> > >>> ----------------------------- Statistics Screen ------- [1-9]: > >>> Change > >>> Screen -- > >>> Start Time | 2009-02-27 > >>> 09:11:31 > >>> Last Reset Time | 2009-03-02 > >>> 07:49:10 > >>> Current Time | 2009-03-02 > >>> 07:49:11 > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Counter Name | Periodic value | Cumulative > >>> value > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Elapsed Time | 00:00:00:857 | > >>> 70:37:39:429 > >>> Call Rate | 0.000 cps | 20.739 > >>> cps > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Incoming call created | 0 | > >>> 0 > >>> OutGoing call created | 0 | > >>> 5273022 > >>> Total Call created | | > >>> 5273022 > >>> Current Call | 34 > >>> | > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Successful call | 0 | > >>> 5271434 > >>> Failed call | 0 | > >>> 1554 > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Response Time 1 | 00:00:00:000 | > >>> 00:00:00:240 > >>> Call Length | 38:32:13:386 | > >>> 00:00:36:131 > >>> ------------------------------ Test Terminated > >>> -------------------------------- > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mszlazak at aol.com Tue Mar 3 23:24:10 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 04 Mar 2009 02:24:10 -0500 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net><57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org><49ADD3E4.20408@gmx.net> <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> Message-ID: <8CB6AB158D38B9A-A74-24FC@WEBMAIL-MA07.sysops.aol.com> Brian, Peter says: "mod_pockesphinx has changed/evolved significantely" Since this seems to be coming without any warning, what specifically are all these and future changes and why are they happening? Mark. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tue, 3 Mar 2009 7:00 pm Subject: Re: [Freeswitch-users] pocketsphinx and event socket Well you should use ESL then ;) /b On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote: > Thank you Brian, > > I will try this later. > > Currently I was happy to get this working on SVN 10003. > > As : Will there also > be major changes in the events I receive through mod_eventsocket? > I spend some time on parsing the right data out of the eventsocket > interface, and I would just have an idea, if I will have to expect > significant work to do, when I later switch to the current SVN. > > Will I need updated grammar files for the other models too? > > Best regards > Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/57128233/attachment.html From Claudio.Cavalera at italtel.it Wed Mar 4 03:41:53 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Mar 2009 12:41:53 +0100 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > freeswitch-users-bounces at lists.freeswitch.org wrote: >> Cavalera Claudio Luigi wrote: >>> So could you please someone more expert with me clarify the >>> difference between -d and -l options so that I can update the wiki >>> which is now wrong/incomplete ? >>> >>> -l, --loglevel=command Log Level >>> -q, --quiet Disable logging >>> -d, --debug=level Debug Level (0 - 7) >> >> The difference is that -d controls the level of debugging output >> generated by fs_cli itself. The log level controls which log messages >> from your running FreeSWITCH daemon are printed to the fs_cli >> console. Could you please explain why if I issue this command in fs_cli: "fsctl loglevel warning" to lower the log level in output file freeswitch.log but then I can't have level debug in fs_cli even if I re-set it with "/log debug" The fsctl loglevel (which I guess is the same as the global loglevel in switch.conf.xml) must always be higher than loglevel in fs_cli? If correct, is this true also for the console_loglevel ? Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From helmut.kuper at ewetel.de Wed Mar 4 04:28:52 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 04 Mar 2009 13:28:52 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> Message-ID: <49AE7404.8000905@ewetel.de> Hi, well, both extensions are direct connected to FS, so we have an internal call. Caller is snom 370, callee is snom 820, codecs are G.722 with RTP crypt AES 32, both phones are within the same LAN. FS is in a separate LAN. Energy levels for speakers and micros on both snom phones are the same. I use latest SVN trunk. regards Helmut On 03.03.2009 16:40, Brian West wrote: > Thats going to depend on a lot of issues... are you on SVN trunk? > What codecs? What is the path? > > /b > From mrene_lists at avgs.ca Wed Mar 4 04:49:06 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 07:49:06 -0500 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: References: Message-ID: <0321F1A5-F307-465E-8204-FE9FF4842900@avgs.ca> fsctl loglevel [xxx] tells the core to ignore log messages not at least that level. They wont get logged, you wont see them on the console and no events will be generated. console loglevel [xxx] (on the "real" console) tells mod_console to do that filtering. /log [xxx] tells fs_cli to do the filtering. It all depends where you filter it, if you want to change the logfile's level without affecting anything else, I recommand you edit logfile.conf.xml Math On 4-Mar-09, at 6:41 AM, Cavalera Claudio Luigi wrote: > freeswitch-users-bounces at lists.freeswitch.org wrote: >> freeswitch-users-bounces at lists.freeswitch.org wrote: >>> Cavalera Claudio Luigi wrote: >>>> So could you please someone more expert with me clarify the >>>> difference between -d and -l options so that I can update the wiki >>>> which is now wrong/incomplete ? >>>> >>>> -l, --loglevel=command Log Level >>>> -q, --quiet Disable logging >>>> -d, --debug=level Debug Level (0 - 7) >>> >>> The difference is that -d controls the level of debugging output >>> generated by fs_cli itself. The log level controls which log >>> messages >>> from your running FreeSWITCH daemon are printed to the fs_cli >>> console. > > > Could you please explain why if I issue this command in fs_cli: > "fsctl loglevel warning" > to lower the log level in output file freeswitch.log > but then I can't have level debug in fs_cli even if I re-set it with > "/log debug" > > The fsctl loglevel (which I guess is the same as the global loglevel > in > switch.conf.xml) must always be higher than loglevel in fs_cli? > If correct, is this true also for the console_loglevel ? > > Thanks, > Claudio > > > 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-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Wed Mar 4 05:08:19 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 04 Mar 2009 14:08:19 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49AE7404.8000905@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> Message-ID: <49AE7D43.9070409@ewetel.de> Hello, I just did a further test to nail down the volume problem with recording a call. Caller as well as callee recorded the session in stereo. I got two files in "recordings". In both files, the party which starts the recording seems to be louder and more clear as the opposite party. It seems to me, that the party which didn't start the recording was transcoded to a different codec (guess g711). Remember: Both partys are using g722 which was confirmed by the phone's display. regards helmut On 04.03.2009 13:28, Helmut Kuper wrote: > Hi, > > well, both extensions are direct connected to FS, so we have an internal > call. Caller is snom 370, callee is snom 820, codecs are G.722 with RTP > crypt AES 32, both phones are within the same LAN. FS is in a separate > LAN. Energy levels for speakers and micros on both snom phones are the same. > > I use latest SVN trunk. > > regards > Helmut From Claudio.Cavalera at italtel.it Wed Mar 4 07:03:46 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Mar 2009 16:03:46 +0100 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: <0321F1A5-F307-465E-8204-FE9FF4842900@avgs.ca> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > fsctl loglevel [xxx] tells the core to ignore log messages not at > least that level. They wont get logged, you wont see them on the > console and no events will be generated. > > console loglevel [xxx] (on the "real" console) tells > mod_console to do > that filtering. > /log [xxx] tells fs_cli to do the filtering. > > It all depends where you filter it, if you want to change the > logfile's level without affecting anything else, I recommand > you edit > logfile.conf.xml > > Math Ah ok, so it's better to keep switch.conf.xml to debug and then to adjust the single sources to lower logging levels. I have also put these lines in a rc.freeswitch script before starting freeswitch export SOFIA_DEBUG=9 export NUA_DEBUG=9 export NTA_DEBUG=9 export TPORT_DEBUG=9 export TPORT_LOG=1 becase I wanted to have all the sofia output redirected to a different logfile but it does not work. Instead if on any console I use fs_cli I see there the sofia stuff, how does that work? Thanks, Claudio Internet E. Mail Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. From brian at freeswitch.org Wed Mar 4 07:13:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Mar 2009 09:13:01 -0600 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49AE7D43.9070409@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> <49AE7D43.9070409@ewetel.de> Message-ID: <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> Can you email me a sample recording please? /b On Mar 4, 2009, at 7:08 AM, Helmut Kuper wrote: > Hello, > > I just did a further test to nail down the volume problem with > recording > a call. Caller as well as callee recorded the session in stereo. I got > two files in "recordings". In both files, the party which starts the > recording seems to be louder and more clear as the opposite party. It > seems to me, that the party which didn't start the recording was > transcoded to a different codec (guess g711). Remember: Both partys > are > using g722 which was confirmed by the phone's display. > > > regards > helmut From helmut.kuper at ewetel.de Wed Mar 4 07:37:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 04 Mar 2009 16:37:38 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> <49AE7D43.9070409@ewetel.de> <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> Message-ID: <49AEA042.3090908@ewetel.de> Hi Brian, both files are on their way ... Quite big (18MB) regards Helmut On 04.03.2009 16:13, Brian West wrote: > Can you email me a sample recording please? > > /b > > On Mar 4, 2009, at 7:08 AM, Helmut Kuper wrote: > > >> Hello, >> >> I just did a further test to nail down the volume problem with >> recording >> a call. Caller as well as callee recorded the session in stereo. I got >> two files in "recordings". In both files, the party which starts the >> recording seems to be louder and more clear as the opposite party. It >> seems to me, that the party which didn't start the recording was >> transcoded to a different codec (guess g711). Remember: Both partys >> are >> using g722 which was confirmed by the phone's display. >> >> >> regards >> helmut >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/1f04e99b/attachment.html From brian at freeswitch.org Wed Mar 4 07:43:51 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Mar 2009 09:43:51 -0600 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49AEA042.3090908@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> <49AE7D43.9070409@ewetel.de> <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> <49AEA042.3090908@ewetel.de> Message-ID: Already received them.. Opened them in Audacity.. compared them... listened to them a couple of times... and they sound fine to me.... So answer the two questions I asked in the private email and we'll see ;) /b On Mar 4, 2009, at 9:37 AM, Helmut Kuper wrote: > Hi Brian, > > both files are on their way ... Quite big (18MB) > > regards > Helmut From helmut.kuper at ewetel.de Wed Mar 4 08:54:14 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 04 Mar 2009 17:54:14 +0100 Subject: [Freeswitch-users] How to send CUSTOM event via event socket? Message-ID: <49AEB236.10201@ewetel.de> Hello, how to send a CUSTOM event via event socket? Currently I send this: sendevent CUSTOM\n Event-Name: CUSTOM\n Event-Subclass: myevent::snom-aurl\n Snom-Event: OFFHOOK\n \n I got an +OK reply from FS. I subscribed to ALL event and get only this: Content-Length: 516 Content-Type: text/event-plain Event-Name: COMMAND Core-UUID: 7b2b75a4-08ca-11de-b1f3-3d0cddd2708d FreeSWITCH-Hostname: ippbx-prod-node0 FreeSWITCH-IPv4: 85.16.246.5 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-03-04%2017%3A42%3A35 Event-Date-GMT: Wed,%2004%20Mar%202009%2016%3A42%3A35%20GMT Event-Date-Timestamp: 1236184955199959 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1045 Command: sendevent%20CUSTOM Event-Name: CUSTOM Event-Subclass: ewetel%3A%3Asnom-aurl Snom-Event: OFFHOOK But there is no CUSTOM event received. regards helmut From fax at virgintechnologies.com Wed Mar 4 10:45:45 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Wed, 04 Mar 2009 18:45:45 +0000 Subject: [Freeswitch-users] separate gateways SIP and RTP stream Message-ID: I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060. Thank you Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/c20f8a51/attachment.html From mrene_lists at avgs.ca Wed Mar 4 10:47:47 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 13:47:47 -0500 Subject: [Freeswitch-users] separate gateways SIP and RTP stream In-Reply-To: References: Message-ID: <1AE3D5A9-6753-4005-947A-A92B16F55DF9@avgs.ca> Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. Math On 4-Mar-09, at 1:45 PM, Justin Miller wrote: > I'm setting up a test SIP trunk with Allstream. They are using > separate gateway IPs for SIP signalling and RTP streams. Has anyone > done this with Freeswitch? I was planning on setting them up as an > extension in the internal profile since they use SIP port 5060. > Thank you > Justin > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/7182b36e/attachment.html From fax at virgintechnologies.com Wed Mar 4 11:25:03 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Wed, 04 Mar 2009 19:25:03 +0000 Subject: [Freeswitch-users] separate gateways SIP and RTP stream Message-ID: So is there an "RTP_proxy" parameter that can be set for the gateway, or in my case, the user extension? How would I define the separate IP for the RTP stream. -----Original Message----- From: Mathieu Rene [mailto:mrene_lists at avgs.ca] Sent: Wednesday, March 4, 2009 11:47 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. Math On 4-Mar-09, at 1:45 PM, Justin Miller wrote: I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060. Thank you Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/709abda0/attachment-0001.html From mrene_lists at avgs.ca Wed Mar 4 11:26:10 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 14:26:10 -0500 Subject: [Freeswitch-users] separate gateways SIP and RTP stream In-Reply-To: References: Message-ID: <259504C8-C9CA-4A0E-98E6-DC091734D356@avgs.ca> You don't. The SIP protocol already takes care of that, just make sure you open your firewall properly. On 4-Mar-09, at 2:25 PM, Justin Miller wrote: > So is there an "RTP_proxy" parameter that can be set for the > gateway, or in my case, the user extension? How would I define the > separate IP for the RTP stream. > -----Original Message----- > From: Mathieu Rene [mailto:mrene_lists at avgs.ca] > Sent: Wednesday, March 4, 2009 11:47 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Using different IPs for signalling and media is perfectly legal, > thats why the SDP contains information on where the media should be > sent.. > > > Math > > On 4-Mar-09, at 1:45 PM, Justin Miller wrote: > >> I'm setting up a test SIP trunk with Allstream. They are using >> separate gateway IPs for SIP signalling and RTP streams. Has >> anyone done this with Freeswitch? I was planning on setting them >> up as an extension in the internal profile since they use SIP port >> 5060. >> Thank you >> Justin >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/064d6a11/attachment.html From mrene_lists at avgs.ca Wed Mar 4 11:27:43 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 14:27:43 -0500 Subject: [Freeswitch-users] How to send CUSTOM event via event socket? In-Reply-To: <49AEB236.10201@ewetel.de> References: <49AEB236.10201@ewetel.de> Message-ID: <0C5EA2F5-9B6D-4D00-98D1-2DD60C8E8F59@avgs.ca> On 4-Mar-09, at 11:54 AM, Helmut Kuper wrote: > Event-Name: CUSTOM > Event-Subclass: ewetel%3A%3Asnom-aurl > Snom-Event: OFFHOOK ^^ mod_event_socket loads up commands it receives in an event struct, when you do a "sendevent", it changes the eventid to be changed to the one you specified, unfortunately it doesnt remove the first Event-Name header. Open a JIRA if thats what you want. Math > Hello, > > how to send a CUSTOM event via event socket? Currently I send this: > > sendevent CUSTOM\n > Event-Name: CUSTOM\n > Event-Subclass: myevent::snom-aurl\n > Snom-Event: OFFHOOK\n > \n > > > I got an +OK reply from FS. I subscribed to ALL event and get only > this: > > Content-Length: 516 > Content-Type: text/event-plain > > Event-Name: COMMAND > Core-UUID: 7b2b75a4-08ca-11de-b1f3-3d0cddd2708d > FreeSWITCH-Hostname: ippbx-prod-node0 > FreeSWITCH-IPv4: 85.16.246.5 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-03-04%2017%3A42%3A35 > Event-Date-GMT: Wed,%2004%20Mar%202009%2016%3A42%3A35%20GMT > Event-Date-Timestamp: 1236184955199959 > Event-Calling-File: mod_event_socket.c > Event-Calling-Function: read_packet > Event-Calling-Line-Number: 1045 > Command: sendevent%20CUSTOM > Event-Name: CUSTOM > Event-Subclass: ewetel%3A%3Asnom-aurl > Snom-Event: OFFHOOK > > But there is no CUSTOM event received. > > regards > helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fax at virgintechnologies.com Wed Mar 4 12:25:16 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Wed, 04 Mar 2009 20:25:16 +0000 Subject: [Freeswitch-users] separate gateways SIP and RTP stream Message-ID: Ok, got it. Thank you. -----Original Message----- From: Mathieu Rene [mailto:mrene_lists at avgs.ca] Sent: Wednesday, March 4, 2009 12:26 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream You don't. The SIP protocol already takes care of that, just make sure you open your firewall properly. On 4-Mar-09, at 2:25 PM, Justin Miller wrote: So is there an "RTP_proxy" parameter that can be set for the gateway, or in my case, the user extension? How would I define the separate IP for the RTP stream. -----Original Message----- From: Mathieu Rene [mailto:mrene_lists at avgs.ca] Sent: Wednesday, March 4, 2009 11:47 AM To:freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. Math On 4-Mar-09, at 1:45 PM, Justin Miller wrote: I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060. Thank you Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/f49be49a/attachment.html From dule.maillist at gmail.com Wed Mar 4 15:58:19 2009 From: dule.maillist at gmail.com (Dan Le) Date: Wed, 4 Mar 2009 18:58:19 -0500 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> Message-ID: <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com> What does do exactly? When do you need it? I'll wikify the response. Thanks, Dan On Tue, Feb 10, 2009 at 8:48 PM, Brian West wrote: > I highly recommend you wipe the box/install and install from Scratch > using SVN trunk > > > /b > - Show quoted text - > > On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote: > > > I'm not able to find any documentation on this setting. I think it may > > be newer than my version of FreeSwitch (1.0). What does it do? > > > > Thanks, > > - Jesse > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/880a76ac/attachment.html From mrene_lists at avgs.ca Wed Mar 4 16:00:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 19:00:33 -0500 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com> Message-ID: <226909BF-C3DC-4172-803A-34329C31CC63@avgs.ca> it auto restarts the profile when a network address change is detected. On 4-Mar-09, at 6:58 PM, Dan Le wrote: > What does do exactly? > When do you need it? > > I'll wikify the response. > > Thanks, > Dan > > On Tue, Feb 10, 2009 at 8:48 PM, Brian West > wrote: > I highly recommend you wipe the box/install and install from Scratch > using SVN trunk > > > /b > - Show quoted text - > > On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote: > > > I'm not able to find any documentation on this setting. I think it > may > > be newer than my version of FreeSwitch (1.0). What does it do? > > > > Thanks, > > - Jesse > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/8454d3ba/attachment-0001.html From dule.maillist at gmail.com Wed Mar 4 18:54:50 2009 From: dule.maillist at gmail.com (Dan Le) Date: Wed, 4 Mar 2009 21:54:50 -0500 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <226909BF-C3DC-4172-803A-34329C31CC63@avgs.ca> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com> <226909BF-C3DC-4172-803A-34329C31CC63@avgs.ca> Message-ID: <914fc92a0903041854k4f5a95e2j3692ca09c0582309@mail.gmail.com> Thanks, wiki'd as: http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address Dan On Wed, Mar 4, 2009 at 7:00 PM, Mathieu Rene wrote: > it auto restarts the profile when a network address change is detected.- > Show quoted text - > > On 4-Mar-09, at 6:58 PM, Dan Le wrote: > > What does do exactly? When do > you need it? > I'll wikify the response. > > Thanks, > Dan > > On Tue, Feb 10, 2009 at 8:48 PM, Brian West wrote: > >> I highly recommend you wipe the box/install and install from Scratch >> using SVN trunk >> >> >> /b >> - Show quoted text - >> >> On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote: >> >> > I'm not able to find any documentation on this setting. I think it may >> > be newer than my version of FreeSwitch (1.0). What does it do? >> > >> > Thanks, >> > - Jesse >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/1a0e1975/attachment.html From gerry at pstn2.net Wed Mar 4 19:31:24 2009 From: gerry at pstn2.net (Gerry Hull) Date: Wed, 4 Mar 2009 22:31:24 -0500 Subject: [Freeswitch-users] mod_unistim? Message-ID: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> I hear rumors that someone is porting chan_unistim to mod_unistim for FreeSwitch?? I hope so -- I use this on my Asterisk box and would love to use it with FS. There are TONS of i2004 phones on the surplus market these days... I've been buying NOS i2004s, virgin, for less than $10US... The full duplex speakerphones in these phones are as or better than a Polycom. I'll be happy to test this module -- I'm just not a C/++ guy. Gerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/972e1db3/attachment.html From mike at jerris.com Wed Mar 4 19:55:59 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 4 Mar 2009 22:55:59 -0500 Subject: [Freeswitch-users] mod_unistim? In-Reply-To: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> References: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> Message-ID: Due to licensing reasons, you can not "port" a gpl piece of code to FreeSWITCH due to restrictions imposed by the gpl so it is not possible to do this unless all copy-write holders approve a license change. Mike On Mar 4, 2009, at 10:31 PM, Gerry Hull wrote: > I hear rumors that someone is porting chan_unistim to mod_unistim > for FreeSwitch?? I hope so -- I use this on my Asterisk box and > would love to use it with FS. There are TONS of i2004 phones on > the surplus market these days... I've been buying NOS i2004s, > virgin, for less than $10US... The full duplex speakerphones in > these phones are as or better than a Polycom. I'll be happy to > test this module -- I'm just not a C/++ guy. > > Gerry > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Mar 4 20:00:40 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Mar 2009 22:00:40 -0600 Subject: [Freeswitch-users] mod_unistim? In-Reply-To: References: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> Message-ID: Actually in this case you can we were giving FULL rights to do what we wanted with the code from the original author. ;) I still have the emails about it.. and someone asked me about this a few weeks ago. /b On Mar 4, 2009, at 9:55 PM, Michael Jerris wrote: > Due to licensing reasons, you can not "port" a gpl piece of code to > FreeSWITCH due to restrictions imposed by the gpl so it is not > possible to do this unless all copy-write holders approve a license > change. > > Mike From kawarod at laposte.net Thu Mar 5 04:12:51 2009 From: kawarod at laposte.net (rod) Date: Thu, 05 Mar 2009 16:12:51 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID Message-ID: <49AFC1C3.9030603@laposte.net> Dear list, I'd like to rewrite the number in the Remote Party ID header and only in this header. ex: I'd like to prefix the caller ID with a prefix code (000 in this example) in the RPID header : From: Anonymous;tag=1208367 Remote-Party-ID: ;privacy=full;screen=yes;party=calling should become: From: Anonymous;tag=1208367 Remote-Party-ID: ;privacy=full;screen=yes;party=calling But the From field should remain unchanged. And how to strip this prefix: From: Anonymous;tag=1208367 Remote-Party-ID: ;privacy=full;screen=yes;party=calling should become: From: Anonymous;tag=1208367 Remote-Party-ID: ;privacy=full;screen=yes;party=calling regards. From brian at freeswitch.org Thu Mar 5 04:23:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 06:23:02 -0600 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49AFC1C3.9030603@laposte.net> References: <49AFC1C3.9030603@laposte.net> Message-ID: <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> Well this depends on how you're placing the call.. if its a standard bridge you can on the A-Leg set "effective_caller_id_number=000$ {caller_id_number}" before you call bridge. Is the from already in the correct format? /b On Mar 5, 2009, at 6:12 AM, rod wrote: > Dear list, > > I'd like to rewrite the number in the Remote Party ID header and > only in > this header. > > ex: I'd like to prefix the caller ID with a prefix code (000 in this > example) in the RPID header : > > From: Anonymous;tag=1208367 > Remote-Party-ID: > 123456 > @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling > > should become: > From: Anonymous;tag=1208367 > Remote-Party-ID: > 000123456 > @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling > > But the From field should remain unchanged. > > And how to strip this prefix: > From: Anonymous;tag=1208367 > Remote-Party-ID: > 000123456 > @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling > > should become: > From: Anonymous;tag=1208367 > Remote-Party-ID: > 123456 > @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling > > > regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/bd880b43/attachment.html From kawarod at laposte.net Thu Mar 5 04:46:51 2009 From: kawarod at laposte.net (rod) Date: Thu, 05 Mar 2009 16:46:51 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> Message-ID: <49AFC9BB.9090106@laposte.net> Hi Brian, if I use the function effective_caller_id_number with my INVITE, I get this: From: "Anonymous" ;tag=17geyFjX5p0gS. this is not exactly what I'm looking for :p rod Brian West wrote: > Well this depends on how you're placing the call.. if its a standard > bridge you can on the A-Leg set > "effective_caller_id_number=000${caller_id_number}" before you call > bridge. > > Is the from already in the correct format? > > /b > > On Mar 5, 2009, at 6:12 AM, rod wrote: > >> Dear list, >> >> I'd like to rewrite the number in the Remote Party ID header and only in >> this header. >> >> ex: I'd like to prefix the caller ID with a prefix code (000 in this >> example) in the RPID header : >> >> From: Anonymous;tag=1208367 >> Remote-Party-ID: >> ;privacy=full;screen=yes;party=calling >> >> should become: >> From: Anonymous;tag=1208367 >> Remote-Party-ID: >> ;privacy=full;screen=yes;party=calling >> >> But the From field should remain unchanged. >> >> And how to strip this prefix: >> From: Anonymous;tag=1208367 >> Remote-Party-ID: >> ;privacy=full;screen=yes;party=calling >> >> should become: >> From: Anonymous;tag=1208367 >> Remote-Party-ID: >> ;privacy=full;screen=yes;party=calling >> >> >> regards. > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kawarod at laposte.net Thu Mar 5 05:00:50 2009 From: kawarod at laposte.net (rod) Date: Thu, 05 Mar 2009 17:00:50 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49AFC9BB.9090106@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> Message-ID: <49AFCD02.2000603@laposte.net> the A leg invite looks like this: From: "Anonymous" it has been rewritten like this: From: "Anonymous" rod rod wrote: > Hi Brian, > > if I use the function effective_caller_id_number with my INVITE, I get this: > > From: "Anonymous" ;tag=17geyFjX5p0gS. > > this is not exactly what I'm looking for :p > > rod > > > Brian West wrote: > >> Well this depends on how you're placing the call.. if its a standard >> bridge you can on the A-Leg set >> "effective_caller_id_number=000${caller_id_number}" before you call >> bridge. >> >> Is the from already in the correct format? >> >> /b >> >> On Mar 5, 2009, at 6:12 AM, rod wrote: >> >> >>> Dear list, >>> >>> I'd like to rewrite the number in the Remote Party ID header and only in >>> this header. >>> >>> ex: I'd like to prefix the caller ID with a prefix code (000 in this >>> example) in the RPID header : >>> >>> From: Anonymous;tag=1208367 >>> Remote-Party-ID: >>> ;privacy=full;screen=yes;party=calling >>> >>> should become: >>> From: Anonymous;tag=1208367 >>> Remote-Party-ID: >>> ;privacy=full;screen=yes;party=calling >>> >>> But the From field should remain unchanged. >>> >>> And how to strip this prefix: >>> From: Anonymous;tag=1208367 >>> Remote-Party-ID: >>> ;privacy=full;screen=yes;party=calling >>> >>> should become: >>> From: Anonymous;tag=1208367 >>> Remote-Party-ID: >>> ;privacy=full;screen=yes;party=calling >>> >>> >>> regards. >>> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From gibbedhead at gmail.com Thu Mar 5 02:14:57 2009 From: gibbedhead at gmail.com (J Mann/Harry) Date: Thu, 5 Mar 2009 05:14:57 -0500 Subject: [Freeswitch-users] Please end the torment Message-ID: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> No, I've yet to contribute anything, I barely have my system doing what I want. But I REALLY love Freeswitch and I want to see it BURY Asterisk. (Windows server user here) I've been struggling with the XML configs, trying to figure out what does what and where! That's fine, I'm used to it. What I'm NOT used to is the total lack of a forum-based community to join and participate in! Where can users SHARE their configs, help each other, learn from each others mistakes? No DEV forum? I'm speechless. Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" at best! I want stickies, a forum for noobs, converts, a dev forum... So on... "WELCOME To FreeSWITCH!" Am I asking too much here? A FORUM? I can't see how you can spark interest when we're so sorely lacking the most basic and widely used community environments on the net! HELP SOMEBODY? Install SMF ASAP! http://www.simplemachines.org BTW: I hate people who voice problems without offering viable solutions in the process... Disgusting! So if someone can offer up a simple hosting account, Control Panel 10, Windows Linux whatever... I'll be more than happy to have SMF setup and receiving user registrations within 24 hours! I've done it dozens of times before. I will then gladly turn over the keys to the kingdom to the "powers that be" and take a backseat, being simply a happy user from that point on! Please folks, please. I'm dying over here and I'm sure I'm not alone! I'm searching Google and finding nothing!! FORUMS! Harry (email me here) switchserver at gmail.com (my FS email) From mrene_lists at avgs.ca Thu Mar 5 06:04:15 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Mar 2009 09:04:15 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> Message-ID: The wiki explains all that and allows all that, whats wrong with IRC? Come and ask questions if you don't understand, you'll get your answers quicker than complaining about the lack of forum. Math On 5-Mar-09, at 5:14 AM, J Mann/Harry wrote: > No, I've yet to contribute anything, I barely have my system doing > what I want. But I REALLY love Freeswitch and I want to see it BURY > Asterisk. (Windows server user here) > > I've been struggling with the XML configs, trying to figure out what > does what and where! That's fine, I'm used to it. What I'm NOT used to > is the total lack of a forum-based community to join and participate > in! Where can users SHARE their configs, help each other, learn from > each others mistakes? No DEV forum? I'm speechless. > > Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" > at best! I want stickies, a forum for noobs, converts, a dev forum... > So on... > > "WELCOME To FreeSWITCH!" > > Am I asking too much here? A FORUM? > > I can't see how you can spark interest when we're so sorely lacking > the most basic and widely used community environments on the net! > > HELP SOMEBODY? Install SMF ASAP! > http://www.simplemachines.org > > BTW: I hate people who voice problems without offering viable > solutions in the process... Disgusting! So if someone can offer up a > simple hosting account, Control Panel 10, Windows Linux whatever... > I'll be more than happy to have SMF setup and receiving user > registrations within 24 hours! I've done it dozens of times before. I > will then gladly turn over the keys to the kingdom to the "powers that > be" and take a backseat, being simply a happy user from that point on! > > Please folks, please. I'm dying over here and I'm sure I'm not alone! > I'm searching Google and finding nothing!! FORUMS! > > Harry (email me here) > switchserver at gmail.com (my FS email) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at suspicious.org Thu Mar 5 06:11:19 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 05 Mar 2009 08:11:19 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: Message-ID: Not to mention there is a full archive of the mailing list on nabble http://www.nabble.com/Freeswitch-users-f32209.html > From: Mathieu Rene > Reply-To: > Date: Thu, 5 Mar 2009 09:04:15 -0500 > To: > Subject: Re: [Freeswitch-users] Please end the torment > > The wiki explains all that and allows all that, whats wrong with IRC? > Come and ask questions if you don't understand, you'll get your > answers quicker than complaining about the lack of forum. > > Math > > On 5-Mar-09, at 5:14 AM, J Mann/Harry wrote: > >> No, I've yet to contribute anything, I barely have my system doing >> what I want. But I REALLY love Freeswitch and I want to see it BURY >> Asterisk. (Windows server user here) >> >> I've been struggling with the XML configs, trying to figure out what >> does what and where! That's fine, I'm used to it. What I'm NOT used to >> is the total lack of a forum-based community to join and participate >> in! Where can users SHARE their configs, help each other, learn from >> each others mistakes? No DEV forum? I'm speechless. >> >> Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" >> at best! I want stickies, a forum for noobs, converts, a dev forum... >> So on... >> >> "WELCOME To FreeSWITCH!" >> >> Am I asking too much here? A FORUM? >> >> I can't see how you can spark interest when we're so sorely lacking >> the most basic and widely used community environments on the net! >> >> HELP SOMEBODY? Install SMF ASAP! >> http://www.simplemachines.org >> >> BTW: I hate people who voice problems without offering viable >> solutions in the process... Disgusting! So if someone can offer up a >> simple hosting account, Control Panel 10, Windows Linux whatever... >> I'll be more than happy to have SMF setup and receiving user >> registrations within 24 hours! I've done it dozens of times before. I >> will then gladly turn over the keys to the kingdom to the "powers that >> be" and take a backseat, being simply a happy user from that point on! >> >> Please folks, please. I'm dying over here and I'm sure I'm not alone! >> I'm searching Google and finding nothing!! FORUMS! >> >> Harry (email me here) >> switchserver at gmail.com (my FS email) >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Mar 5 06:22:32 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Mar 2009 06:22:32 -0800 (PST) Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> Message-ID: <1236262952915-2429661.post@n2.nabble.com> Much more than an archive, nabble makes a full embeddable forum that is linked to the mailing list. We will be embedding this in our webpage soon for the best of both worlds, a forum and a mailing list without the additional overhead of having to monitor 2 things. Mike Ken Rice-3 wrote: > > Not to mention there is a full archive of the mailing list on nabble > > http://www.nabble.com/Freeswitch-users-f32209.html > > -- View this message in context: http://n2.nabble.com/Please-end-the-torment-tp2429589p2429661.html Sent from the freeswitch-users mailing list archive at Nabble.com. From damin at nacs.net Thu Mar 5 06:46:44 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 5 Mar 2009 09:46:44 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: Message-ID: <123301c99da1$33ee98e0$9bcbcaa0$@net> I hate forums. Forums suck. They are a pain in the ass to search and find things. I prefer the mailing list and the Wiki. Can we please keep it that way? It is really easy to find stuff via Nabble and the Wiki. From gmaruzz at celliax.org Thu Mar 5 07:11:25 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 5 Mar 2009 16:11:25 +0100 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <1236262952915-2429661.post@n2.nabble.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <1236262952915-2429661.post@n2.nabble.com> Message-ID: <7b197bef0903050711v5a8d0eefo31c9e476fe6167ae@mail.gmail.com> On Thu, Mar 5, 2009 at 3:22 PM, Michael Jerris wrote: > > Much more than an archive, nabble makes a full embeddable forum that is > linked to the mailing list. ?We will be embedding this in our webpage soon > for the best of both worlds, a forum and a mailing list without the > additional overhead of having to monitor 2 things. agree! From helmut.kuper at ewetel.de Thu Mar 5 07:31:34 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 05 Mar 2009 16:31:34 +0100 Subject: [Freeswitch-users] Patch for mod_event_socket to fire real CUSTOM events via sendevent command Message-ID: <49AFF056.70706@ewetel.de> Hello, I enhanced mod_event_socket's sendevent command to fire real CUSTOM events with correct subclass and trailing custom headers, so that subscribed nodes can receive those events. Is there a way to get the patch into trunk? regards Helmut From csorlie at teldio.com Thu Mar 5 07:41:28 2009 From: csorlie at teldio.com (Cameron Sorlie) Date: Thu, 5 Mar 2009 10:41:28 -0500 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD Message-ID: In a sense, you might say I did futz with mod_vmd ... to create mod_vad. There appeared to be just no (easy) way to modify the internal VAD code in the FreeSWITCH core (see switch_rtp.c) to identify the origins of voice activity. And rather than build into mod_vmd, which is a special purpose tool, a separate module for VAD seemed like a reasonable idea. In short, the mod_vad which I've written up independently monitors the read and the write legs of the session it is run on, and tags each VAD_TALK and VAD_NOTALK event it fires with a user-supplied identification marker (a short string) for the leg which the event relates to. At the moment, the VAD algorithm is dead simple, and is much like the one in the core. I will be happy to submit this module, in a little while, after I've had a chance to make it perhaps a bit more useable outside of our own application context. Cam On Mon, Mar 2, 2009 at 5:43 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > > On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale > wrote: > > i think that's what mod_vmd does > > > I think that's right. It just does the opposite - instead of looking > for differing power levels it looks for the same power level. In other > words it tries to detect distinctly non-human sound. I'll bet you > could futz with that code and have it fire off events when it detects > what it believes is human speech. > > -MC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/f240a1b3/attachment-0001.html From mrene_lists at avgs.ca Thu Mar 5 07:41:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Mar 2009 10:41:33 -0500 Subject: [Freeswitch-users] Patch for mod_event_socket to fire real CUSTOM events via sendevent command In-Reply-To: <49AFF056.70706@ewetel.de> References: <49AFF056.70706@ewetel.de> Message-ID: <8BC75FF7-C82D-49E4-BFEA-E40887372A57@avgs.ca> Sure, open a JIRA as a improvement, and prefix your bug name with [patch] http://jira.freeswitch.org/ Math On 5-Mar-09, at 10:31 AM, Helmut Kuper wrote: > Hello, > > I enhanced mod_event_socket's sendevent command to fire real CUSTOM > events with correct subclass and trailing custom headers, so that > subscribed nodes can receive those events. Is there a way to get the > patch into trunk? > > regards > Helmut > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Mar 5 07:51:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 09:51:15 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: Message-ID: <9AE4AAC4-34A3-4758-B64E-38B0056C5F30@freeswitch.org> Kewl you can contribute via http://jira.freeswitch.org ;) /b On Mar 5, 2009, at 9:41 AM, Cameron Sorlie wrote: > In a sense, you might say I did futz with mod_vmd ... to create > mod_vad. There appeared to be just no (easy) way to modify the > internal VAD code in the FreeSWITCH core (see switch_rtp.c) to > identify the origins of voice activity. And rather than build into > mod_vmd, which is a special purpose tool, a separate module for VAD > seemed like a reasonable idea. > > In short, the mod_vad which I've written up independently monitors > the read and the write legs of the session it is run on, and tags > each VAD_TALK and VAD_NOTALK event it fires with a user-supplied > identification marker (a short string) for the leg which the event > relates to. At the moment, the VAD algorithm is dead simple, and is > much like the one in the core. I will be happy to submit this > module, in a little while, after I've had a chance to make it > perhaps a bit more useable outside of our own application context. > > Cam From helmut.kuper at ewetel.de Thu Mar 5 07:52:27 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 05 Mar 2009 16:52:27 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> <49AE7D43.9070409@ewetel.de> <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> <49AEA042.3090908@ewetel.de> Message-ID: <49AFF53B.5000802@ewetel.de> Hello Brian, I checked the files without earphones and with Audacity. You are right Brian, everything is fine. Sorry for the inconvenience. Thx for your patience. regards helmut From BenHoltsclaw at averyschools.net Thu Mar 5 09:03:50 2009 From: BenHoltsclaw at averyschools.net (Ben Holtsclaw) Date: Thu, 05 Mar 2009 12:03:50 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> Message-ID: <49AFBFA6.45B7.0079.0@averyschools.net> I agree with Harry. I do not like the mailing list. Those that do like the mailing list always advocate Nabble. For those that advocate that solution, do you even realize that you can't post on Nabble unless you are subscribed to the mailing list? I am also not a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client at the same time I uninstalled my NNTP reader. Most of the time, I actually find it difficult to obtain support in the #freeswitch channel. Once you ask the question, if somebody doesn't happen to be there that knows the answer, then you're screwed. How many times have I asked a question only to wait 30 seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the question again? I have found the conversation in #openzap to be much more focused. Thank goodness I'm using that module! In that channel, I never see conversations about cd burners, somebody's girlfriend in South America, or off color jokes about someone's sexual proclivity. And because I know I'll get flamed for saying that, just look at this: [23:10] <{tasker}> me, too, but i'm a different animal [23:10] <{tasker}> in NY and in Miami i went nutz [23:10] lol [23:10] * jefferai is now known as lollerai [23:10] yeah i love her [23:10] <{tasker}> latinas everywhere [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left #freeswitch [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed out) ) [23:11] here its blond blue eyed girls [23:11] * lollerai is now known as lolferai [23:11] brazilians... hopefully she's hot. i've seen some pretty dodgy looking chicks from there [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, "yeaaaaaaaaah, it's just for a few days" [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch [23:11] <{tasker}> blonde / blue eyes are overrated [23:11] <{tasker}> give me a latina any day [23:11] best thing around here though If I'm going into #freeswitch at 11pm at night, it's probably because I really need some help with some problem I've run into after hours. Can you imagine me injecting a question about a SIP profile into that conversation?? ALL that aside... I'm willing to use a carrier pigeon if that's the way the three primary developers wish to communicate. They have been instrumental in getting my project where it is today. You know the saying... beggars can't be choosers. Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x2301 >>> On 3/5/2009 at 5:14 AM, "J Mann/Harry" wrote: No, I've yet to contribute anything, I barely have my system doing what I want. But I REALLY love Freeswitch and I want to see it BURY Asterisk. (Windows server user here) I've been struggling with the XML configs, trying to figure out what does what and where! That's fine, I'm used to it. What I'm NOT used to is the total lack of a forum-based community to join and participate in! Where can users SHARE their configs, help each other, learn from each others mistakes? No DEV forum? I'm speechless. Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" at best! I want stickies, a forum for noobs, converts, a dev forum... So on... "WELCOME To FreeSWITCH!" Am I asking too much here? A FORUM? I can't see how you can spark interest when we're so sorely lacking the most basic and widely used community environments on the net! HELP SOMEBODY? Install SMF ASAP! http://www.simplemachines.org BTW: I hate people who voice problems without offering viable solutions in the process... Disgusting! So if someone can offer up a simple hosting account, Control Panel 10, Windows Linux whatever... I'll be more than happy to have SMF setup and receiving user registrations within 24 hours! I've done it dozens of times before. I will then gladly turn over the keys to the kingdom to the "powers that be" and take a backseat, being simply a happy user from that point on! Please folks, please. I'm dying over here and I'm sure I'm not alone! I'm searching Google and finding nothing!! FORUMS! Harry (email me here) switchserver at gmail.com (my FS email) _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/783d1083/attachment.html From brian at freeswitch.org Thu Mar 5 09:16:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 11:16:08 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <49AFBFA6.45B7.0079.0@averyschools.net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: I have been trying to push all the social talk into #freeswitch-social to keep #freeswitch on topic.. sometimes after hours in the US it gets a bit off topic. I'm usually alive in the channel till around 11PM+ CST most days. I take questions and answer questions at all hours if I'm awake... I too am guilty of going off topic. /b On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: > I agree with Harry. I do not like the mailing list. Those that do > like the mailing list always advocate Nabble. For those that > advocate that solution, do you even realize that you can't post on > Nabble unless you are subscribed to the mailing list? I am also not > a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in > years! I think I uninstalled my IRC client at the same time I > uninstalled my NNTP reader. Most of the time, I actually find it > difficult to obtain support in the #freeswitch channel. Once you ask > the question, if somebody doesn't happen to be there that knows the > answer, then you're screwed. How many times have I asked a question > only to wait 30 seconds and then see, "anthm has joined > #freeswitch." Crap...do I ask the question again? I have found the > conversation in #openzap to be much more focused. Thank goodness I'm > using that module! In that channel, I never see conversations about > cd burners, somebody's girlfriend in South America, or off color > jokes about someone's sexual proclivity. And because I know I'll get > flamed for saying that, just look at this: > > [23:10] <{tasker}> me, too, but i'm a different animal > [23:10] <{tasker}> in NY and in Miami i went nutz > [23:10] lol > [23:10] * jefferai is now known as lollerai > [23:10] yeah i love her > [23:10] <{tasker}> latinas everywhere > [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left > #freeswitch > [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection > timed out)) > [23:11] here its blond blue eyed girls > [23:11] * lollerai is now known as lolferai > [23:11] brazilians... hopefully she's hot. i've seen some > pretty dodgy looking chicks from there > [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, > "yeaaaaaaaaah, it's just for a few days" > [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though > > If I'm going into #freeswitch at 11pm at night, it's probably > because I really need some help with some problem I've run into > after hours. Can you imagine me injecting a question about a SIP > profile into that conversation?? > > ALL that aside... I'm willing to use a carrier pigeon if that's the > way the three primary developers wish to communicate. They have been > instrumental in getting my project where it is today. You know the > saying... beggars can't be choosers. > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/da576a03/attachment-0001.html From mrene_lists at avgs.ca Thu Mar 5 09:19:31 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Mar 2009 12:19:31 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: The bot even takes people's question and keeps them in list so when someone that knows shows up he can answer it, thats the ~take-a-number option... On 5-Mar-09, at 12:16 PM, Brian West wrote: > I have been trying to push all the social talk into #freeswitch- > social to keep #freeswitch on topic.. sometimes after hours in the > US it gets a bit off topic. I'm usually alive in the channel till > around 11PM+ CST most days. I take questions and answer questions > at all hours if I'm awake... I too am guilty of going off topic. > > /b > > > On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: > >> I agree with Harry. I do not like the mailing list. Those that do >> like the mailing list always advocate Nabble. For those that >> advocate that solution, do you even realize that you can't post on >> Nabble unless you are subscribed to the mailing list? I am also not >> a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in >> years! I think I uninstalled my IRC client at the same time I >> uninstalled my NNTP reader. Most of the time, I actually find it >> difficult to obtain support in the #freeswitch channel. Once you >> ask the question, if somebody doesn't happen to be there that knows >> the answer, then you're screwed. How many times have I asked a >> question only to wait 30 seconds and then see, "anthm has joined >> #freeswitch." Crap...do I ask the question again? I have found the >> conversation in #openzap to be much more focused. Thank goodness >> I'm using that module! In that channel, I never see conversations >> about cd burners, somebody's girlfriend in South America, or off >> color jokes about someone's sexual proclivity. And because I know >> I'll get flamed for saying that, just look at this: >> >> [23:10] <{tasker}> me, too, but i'm a different animal >> [23:10] <{tasker}> in NY and in Miami i went nutz >> [23:10] lol >> [23:10] * jefferai is now known as lollerai >> [23:10] yeah i love her >> [23:10] <{tasker}> latinas everywhere >> [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has >> left #freeswitch >> [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection >> timed out)) >> [23:11] here its blond blue eyed girls >> [23:11] * lollerai is now known as lolferai >> [23:11] brazilians... hopefully she's hot. i've seen some >> pretty dodgy looking chicks from there >> [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, >> "yeaaaaaaaaah, it's just for a few days" >> [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch >> [23:11] <{tasker}> blonde / blue eyes are overrated >> [23:11] <{tasker}> give me a latina any day >> [23:11] best thing around here though >> >> If I'm going into #freeswitch at 11pm at night, it's probably >> because I really need some help with some problem I've run into >> after hours. Can you imagine me injecting a question about a SIP >> profile into that conversation?? >> >> ALL that aside... I'm willing to use a carrier pigeon if that's the >> way the three primary developers wish to communicate. They have >> been instrumental in getting my project where it is today. You know >> the saying... beggars can't be choosers. >> >> >> Ben Holtsclaw >> Network Engineer >> Avery County Schools >> Ph: 828.733.3567 x2301 >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/de361bcb/attachment.html From egghunt at gmail.com Thu Mar 5 09:27:59 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Thu, 5 Mar 2009 14:27:59 -0300 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: Besides #freeswitch-social and #openzap, there are other channels related to the project. List: http://wiki.freeswitch.org/wiki/Main_Page#Community_and_Support On Thu, Mar 5, 2009 at 2:19 PM, Mathieu Rene wrote: > The bot even takes people's question and keeps them in list so when someone > that knows shows up he can answer it, thats the ~take-a-number option... > On 5-Mar-09, at 12:16 PM, Brian West wrote: > > I have been trying to push all the social talk into #freeswitch-social to > keep #freeswitch on topic.. sometimes after hours in the US it gets a bit > off topic. I'm usually alive in the channel till around 11PM+ CST most > days. I take questions and answer questions at all hours if I'm awake... I > too am guilty of going off topic. > /b > > > On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: > > I agree with Harry. I do not like the mailing list. Those that do like the > mailing list always advocate Nabble. For those that advocate that solution, > do you even realize that you can't post on Nabble unless you *are* subscribed > to the mailing list? I am also not a fan of IRC. Before I came upon > FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client > at the same time I uninstalled my NNTP reader. Most of the time, I actually > find it difficult to obtain support in the #freeswitch channel. Once you ask > the question, if somebody doesn't happen to be there that knows the answer, > then you're screwed. How many times have I asked a question only to wait 30 > seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the > question again? I *have* found the conversation in #openzap to be much > more focused. Thank goodness I'm using that module! In that channel, I never > see conversations about cd burners, somebody's girlfriend in South America, > or off color jokes about someone's sexual proclivity. And because I know > I'll get flamed for saying that, just look at this: > > [23:10] <{tasker}> me, too, but i'm a different animal > [23:10] <{tasker}> in NY and in Miami i went nutz > [23:10] lol > [23:10] * jefferai is now known as lollerai > [23:10] yeah i love her > [23:10] <{tasker}> latinas everywhere > [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left > #freeswitch > [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed > out) ) > [23:11] here its blond blue eyed girls > [23:11] * lollerai is now known as lolferai > [23:11] brazilians... hopefully she's hot. i've seen some pretty > dodgy looking chicks from there > [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, > "yeaaaaaaaaah, it's just for a few days" > [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though > > If I'm going into #freeswitch at 11pm at night, it's probably because I > really need some help with some problem I've run into after hours. Can you > imagine me injecting a question about a SIP profile into that conversation?? > > ALL that aside... I'm willing to use a carrier pigeon if that's the way the > three primary developers wish to communicate. They have been instrumental in > getting my project where it is today. You know the saying... beggars can't > be choosers. > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Arnaldo M Pereira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/9f412589/attachment.html From BenHoltsclaw at averyschools.net Thu Mar 5 09:27:35 2009 From: BenHoltsclaw at averyschools.net (Ben Holtsclaw) Date: Thu, 05 Mar 2009 12:27:35 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: <49AFC537.45B7.0079.0@averyschools.net> The problem with take-a-number is what if I'm not there when someone can answer it? >>> On 3/5/2009 at 12:19 PM, Mathieu Rene wrote: The bot even takes people's question and keeps them in list so when someone that knows shows up he can answer it, thats the ~take-a-number option... On 5-Mar-09, at 12:16 PM, Brian West wrote: I have been trying to push all the social talk into #freeswitch-social to keep #freeswitch on topic.. sometimes after hours in the US it gets a bit off topic. I'm usually alive in the channel till around 11PM+ CST most days. I take questions and answer questions at all hours if I'm awake... I too am guilty of going off topic. /b On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: I agree with Harry. I do not like the mailing list. Those that do like the mailing list always advocate Nabble. For those that advocate that solution, do you even realize that you can't post on Nabble unless you are subscribed to the mailing list? I am also not a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client at the same time I uninstalled my NNTP reader. Most of the time, I actually find it difficult to obtain support in the #freeswitch channel. Once you ask the question, if somebody doesn't happen to be there that knows the answer, then you're screwed. How many times have I asked a question only to wait 30 seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the question again? I have found the conversation in #openzap to be much more focused. Thank goodness I'm using that module! In that channel, I never see conversations about cd burners, somebody's girlfriend in South America, or off color jokes about someone's sexual proclivity. And because I know I'll get flamed for saying that, just look at this: [23:10] <{tasker}> me, too, but i'm a different animal [23:10] <{tasker}> in NY and in Miami i went nutz [23:10] lol [23:10] * jefferai is now known as lollerai [23:10] yeah i love her [23:10] <{tasker}> latinas everywhere [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left #freeswitch [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed out)) [23:11] here its blond blue eyed girls [23:11] * lollerai is now known as lolferai [23:11] brazilians... hopefully she's hot. i've seen some pretty dodgy looking chicks from there [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, "yeaaaaaaaaah, it's just for a few days" [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch [23:11] <{tasker}> blonde / blue eyes are overrated [23:11] <{tasker}> give me a latina any day [23:11] best thing around here though If I'm going into #freeswitch at 11pm at night, it's probably because I really need some help with some problem I've run into after hours. Can you imagine me injecting a question about a SIP profile into that conversation?? ALL that aside... I'm willing to use a carrier pigeon if that's the way the three primary developers wish to communicate. They have been instrumental in getting my project where it is today. You know the saying... beggars can't be choosers. Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x2301 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/daae766e/attachment-0001.html From gerry at pstn2.net Thu Mar 5 09:30:21 2009 From: gerry at pstn2.net (Gerry Hull) Date: Thu, 5 Mar 2009 12:30:21 -0500 Subject: [Freeswitch-users] mod_unistim? In-Reply-To: References: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> Message-ID: <98a86adf0903050930t7b0df79fm5524a3ec0aaa3b4f@mail.gmail.com> Yeah, I was looking at the Developers IRC log... I see that milkj has given you an MPL license... Has anyone started development on the port? On Wed, Mar 4, 2009 at 11:00 PM, Brian West wrote: > Actually in this case you can we were giving FULL rights to do what we > wanted with the code from the original author. ;) I still have the > emails about it.. and someone asked me about this a few weeks ago. > > /b > > On Mar 4, 2009, at 9:55 PM, Michael Jerris wrote: > > > Due to licensing reasons, you can not "port" a gpl piece of code to > > FreeSWITCH due to restrictions imposed by the gpl so it is not > > possible to do this unless all copy-write holders approve a license > > change. > > > > Mike > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/68c7cea4/attachment.html From mike at jerris.com Thu Mar 5 09:31:14 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Mar 2009 12:31:14 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <49AFBFA6.45B7.0079.0@averyschools.net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: <6F7B8F0B-0975-452D-A6C1-AE484131E2AB@jerris.com> On Mar 5, 2009, at 12:03 PM, Ben Holtsclaw wrote: > I agree with Harry. I do not like the mailing list. Those that do > like the mailing list always advocate Nabble. For those that > advocate that solution, do you even realize that you can't post on > Nabble unless you are subscribed to the mailing list? Fair points, the plan to address this is that we are moving to a new back-end to our hosting infrastructure that has unified logins. Once this is in place we can make it so you are a "member" of the mailing list as soon as you have an id, although set to not receive mail, then we'll have a subscriptions options page where you can just check the lists you want sent to you via email, and the nabble forums can be embedded in our page directly. It is not perfect but much closer to something usable. I understand that some are more comfortable than others with different modes of communication. We are trying to find a balance between being able to serve the community with the best way to communicate for them, and us being able to actually monitor and maintain it. A forum is pretty useless if no one responds or monitors it and already we are flooded by the amount on the mailing list, having to answer the same question the same day in both because someone prefers one to the other and having to spend time checking both is not a good use of time. Do you think that if we better integrate the nabble forums into our site and list subscriptions that it would be a usable system? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/5b2f6de1/attachment.html From anthony.minessale at gmail.com Thu Mar 5 09:33:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Mar 2009 11:33:03 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: <191c3a030903050933p6df4a725o9be751a390f9c916@mail.gmail.com> Ok, SO? We should make a forum? See this one? http://www.voip-info.org/boards/index.php?b=6 Yah, that one has been around for many months and nobody maintains it. We do have a link to it on our homepage but nobody reads it so, there is a dilemma here right? IRC is for sure more useful of a resource between 9am and 6pm GMT-6 in the timezone where me and bkw live. Any other times it would be up to other community members like the few who have been stepping up in a major way in the last few weeks, thank you all. I don't know how good of a job we are doing, but we are a small group and we already spend most of every day managing a bug tracker, irc, and this list. If you took a poll 8/10 people hate forums. I don't really know if i like them or not, I have enjoyed finding forums before that discussed something that happened ages ago that relates to a problem I have and being able to skim months into the future reading the end results, but more times than not they go south and contain a bunch of flaming back and forth about midway through. We already need voulenteers to help manage the resources we already have, Website, Mailing list, JIRA. If we introduce more we, run the risk of doing a bad job at maintaining them. Is anyone willing to help? If you can use nabble to post as long as you have an account on the mailing list, can't you just disable mail dilevery and use the nabble exclusively? If not is there another solution? I really need some help in this department to even consider introducing any more community resources. On Thu, Mar 5, 2009 at 11:16 AM, Brian West wrote: > I have been trying to push all the social talk into #freeswitch-social to > keep #freeswitch on topic.. sometimes after hours in the US it gets a bit > off topic. I'm usually alive in the channel till around 11PM+ CST most > days. I take questions and answer questions at all hours if I'm awake... I > too am guilty of going off topic. > /b > > > On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: > > I agree with Harry. I do not like the mailing list. Those that do like the > mailing list always advocate Nabble. For those that advocate that solution, > do you even realize that you can't post on Nabble unless you *are* subscribed > to the mailing list? I am also not a fan of IRC. Before I came upon > FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client > at the same time I uninstalled my NNTP reader. Most of the time, I actually > find it difficult to obtain support in the #freeswitch channel. Once you ask > the question, if somebody doesn't happen to be there that knows the answer, > then you're screwed. How many times have I asked a question only to wait 30 > seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the > question again? I *have* found the conversation in #openzap to be much > more focused. Thank goodness I'm using that module! In that channel, I never > see conversations about cd burners, somebody's girlfriend in South America, > or off color jokes about someone's sexual proclivity. And because I know > I'll get flamed for saying that, just look at this: > > [23:10] <{tasker}> me, too, but i'm a different animal > [23:10] <{tasker}> in NY and in Miami i went nutz > [23:10] lol > [23:10] * jefferai is now known as lollerai > [23:10] yeah i love her > [23:10] <{tasker}> latinas everywhere > [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left > #freeswitch > [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed > out) ) > [23:11] here its blond blue eyed girls > [23:11] * lollerai is now known as lolferai > [23:11] brazilians... hopefully she's hot. i've seen some pretty > dodgy looking chicks from there > [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, > "yeaaaaaaaaah, it's just for a few days" > [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though > > If I'm going into #freeswitch at 11pm at night, it's probably because I > really need some help with some problem I've run into after hours. Can you > imagine me injecting a question about a SIP profile into that conversation?? > > ALL that aside... I'm willing to use a carrier pigeon if that's the way the > three primary developers wish to communicate. They have been instrumental in > getting my project where it is today. You know the saying... beggars can't > be choosers. > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/2d4752ed/attachment.html From brian at freeswitch.org Thu Mar 5 09:37:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 11:37:36 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <49AFC537.45B7.0079.0@averyschools.net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> Message-ID: <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> Well you can always shoot an email to the mailing list... or wait for someone that can answer it.. which reminds me we really need people to volunteer to help out on the IRC channel so we have skilled people able to answer questions 24/7, I personally cover 12-16 hours a day most of the time. Even weekends! I was even up at 4am today and answered a few. /b On Mar 5, 2009, at 11:27 AM, Ben Holtsclaw wrote: > The problem with take-a-number is what if I'm not there when someone > can answer it? From egghunt at gmail.com Thu Mar 5 09:43:04 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Thu, 5 Mar 2009 14:43:04 -0300 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> Message-ID: On Thu, Mar 5, 2009 at 2:37 PM, Brian West wrote: > Well you can always shoot an email to the mailing list... or wait for > someone that can answer it.. which reminds me we really need people to > volunteer to help out on the IRC channel so we have skilled people > able to answer questions 24/7, I personally cover 12-16 hours a day > most of the time. Even weekends! I was even up at 4am today and > answered a few. I'm usually on the channel starting 9am (GMT -3). Can't answer every question like you guys, but I can surely help. > > > /b > > On Mar 5, 2009, at 11:27 AM, Ben Holtsclaw wrote: > > > The problem with take-a-number is what if I'm not there when someone > > can answer it? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Arnaldo M Pereira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/72df5d9c/attachment-0001.html From damin at nacs.net Thu Mar 5 09:43:43 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 5 Mar 2009 12:43:43 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> Message-ID: <131e01c99db9$ed440ba0$c7cc22e0$@net> You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :) From Prometheus001 at gmx.net Thu Mar 5 12:20:18 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 05 Mar 2009 21:20:18 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> <49ADD3E4.20408@gmx.net> <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> Message-ID: <49B03402.8050601@gmx.net> Hello Brian, concerning > Well you should use ESL then ;) I simply do not understand what you mean by this. Is it sarcastic? Am I asking stupid questions? After upgrading Freeswitch to the newest trunk, mod_pocketsphinx didn't work anymore. So I asked this mailing list about information about what happened. I understand now that there were some significant changes in mod_pocketsphinx and that also some other files have to be updated. I could not find any documentation about these changes, and asking here on this mailing list was rather disappointing for me. Some bits, yes. Some things don't work/crash, as I have read here. We are not using Freeswitch just as a toy to play around. Sometimes it's simply important to know which impact a certain change may have on our system. And other people will run into the same problem. So any advice was needed about the status and how to make it work. I'll update the wiki with this information (as I usually do), I promise. I honor the great work you do and freeswitch is really great. But asking: >Will there also be major changes in the events I receive through mod_eventsocket? >Will I need updated grammar files for the other models too? and receiving > Well you should use ESL then ;) is frustrating. Best regards Peter Brian West schrieb: > Well you should use ESL then ;) > > /b > > On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote: > > >> Thank you Brian, >> >> I will try this later. >> >> Currently I was happy to get this working on SVN 10003. >> >> As mod_pockesphinx has changed/evolved significantely: Will there also >> be major changes in the events I receive through mod_eventsocket? >> I spend some time on parsing the right data out of the eventsocket >> interface, and I would just have an idea, if I will have to expect >> significant work to do, when I later switch to the current SVN. >> >> Will I need updated grammar files for the other models too? >> >> Best regards >> Peter >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu Mar 5 12:32:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Mar 2009 12:32:29 -0800 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49B03402.8050601@gmx.net> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> <49ADD3E4.20408@gmx.net> <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> <49B03402.8050601@gmx.net> Message-ID: <87f2f3b90903051232l22db660blba57c46208f42d63@mail.gmail.com> On Thu, Mar 5, 2009 at 12:20 PM, Peter P GMX wrote: > Hello Brian, > > concerning >> Well you should use ESL then ;) > I simply do not understand what you mean by this. Is it sarcastic? Am I > asking stupid questions? > ESL = Event Socket Library. It is an abstraction layer to make interacting with the FS event socket a little easier. Look in the source directory under libs/esl and you'll see all sorts of stuff. Also check out the new-but-growing ESL wiki page: http://wiki.freeswitch.org/wiki/Esl -MC From kristian.kielhofner at gmail.com Thu Mar 5 12:39:27 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 5 Mar 2009 15:39:27 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <131e01c99db9$ed440ba0$c7cc22e0$@net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> <131e01c99db9$ed440ba0$c7cc22e0$@net> Message-ID: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> A bunch of telephony geeks and a 1900 number - what could go wrong? Anyways, I too don't understand why people prefer forums. I follow dozens of mailling lists and a half a dozen e-mail addresses without ever leaving my mail client. My mail client happens to be gmail, btw: - Much more customization, filtering, etc possible than any "web forum" - "Local" copies of all messages - Search is awesome, ever hear of Google? ;) Web forums are good when you have to serve ads to people to get paid. Other than that they are certainly not the ideal tool for the job. Besides (and don't take this as an insult) - have you ever compared the web forums to the mailing lists for projects that offer both? Say what you want to say about mailing lists and IRC but the reality (usually) is the l33tz all hang out here and web forums (almost always) end up with the same groups of n00bz circling around and around trying to figure out how to accomplish even the most basic of tasks. Obviously that can go both ways but as a rule of thumb the people that are usually in a position to help others typically prefer mailing lists (probably for some of the reasons I cited above). Or maybe they are just old gray hairs too stuck in their ways. I don't know. ;) On Thu, Mar 5, 2009 at 12:43 PM, Gregory Boehnlein wrote: > You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :) > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From mike at jerris.com Thu Mar 5 13:57:30 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Mar 2009 16:57:30 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> <131e01c99db9$ed440ba0$c7cc22e0$@net> <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> Message-ID: <5C6F6B6E-EE30-44CD-996D-ED0A2FD0F894@jerris.com> On Mar 5, 2009, at 3:39 PM, Kristian Kielhofner wrote: > A bunch of telephony geeks and a 1900 number - what could go wrong? > > Anyways, I too don't understand why people prefer forums. > > I follow dozens of mailling lists and a half a dozen e-mail addresses > without ever leaving my mail client. My mail client happens to be > gmail, btw: > > - Much more customization, filtering, etc possible than any "web > forum" > - "Local" copies of all messages > - Search is awesome, ever hear of Google? ;) > > Web forums are good when you have to serve ads to people to get > paid. Other than that they are certainly not the ideal tool for the > job. > > Besides (and don't take this as an insult) - have you ever compared > the web forums to the mailing lists for projects that offer both? Say > what you want to say about mailing lists and IRC but the reality > (usually) is the l33tz all hang out here and web forums (almost > always) end up with the same groups of n00bz circling around and > around trying to figure out how to accomplish even the most basic of > tasks. > > Obviously that can go both ways but as a rule of thumb the people > that are usually in a position to help others typically prefer mailing > lists (probably for some of the reasons I cited above). Or maybe they > are just old gray hairs too stuck in their ways. I don't know. ;) I previously thought the same about the insiders always preferring mailing lists but I have a friend who is a core member of the cacti developers and they apparently prefer the forums to the lists, but have both. Food for thought at least. MIke From dynaguy at gmail.com Thu Mar 5 12:19:49 2009 From: dynaguy at gmail.com (Dyna Guy) Date: Thu, 5 Mar 2009 12:19:49 -0800 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot Message-ID: I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh install) and followed the instruction on the wiki installed the new Freeswitch v1.0.3. After the installation I can start FS by issus command "/usr/local/freeswitch/bin/freeswitch". After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and chmod it to 755. Reboot the server and FS is not running. If I try the start up script: [root at localhost build]# /etc/init.d/freeswitch start Starting freeswitch: [ OK ] [root at localhost build]# /etc/init.d/freeswitch status freeswitch dead but subsys locked Did I miss somthing here? Please help. Thanks a lot. dynaguy Here is the copy of the origenal freeswitch.init.centos5 ---------------------------- [root at localhost ~]# cat /etc/init.d/freeswitch #!/bin/bash # # /etc/rc.d/init.d/freeswitch # # The FreeSwitch Open Source Voice Platform # # chkconfig: 345 89 14 # description: Starts and stops the freeswitch server daemon # processname: freeswitch # config: /usr/local/freeswitch/conf/freeswitch.conf # pidfile: /usr/local/freeswitch/log/freeswitch.pid # # Source function library. . /etc/init.d/functions PROG_NAME=freeswitch PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_USER=${FS_USER-freeswitch} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} LOCK_FILE=/var/lock/subsys/freeswitch FREESWITCH_ARGS="-nc" RETVAL=0 # Source options file if [ -f /etc/sysconfig/freeswitch ]; then . /etc/sysconfig/freeswitch fi # start() { echo -n "Starting $PROG_NAME: " if [ -e $LOCK_FILE ]; then if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then echo echo -n $"$PROG_NAME is already running."; failure $"$PROG_NAME is already running."; echo return 1 fi fi cd $FS_HOME daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" echo RETVAL=$? [ $RETVAL -eq 0 ] && touch $LOCK_FILE; echo return $RETVAL } stop() { echo -n "Shutting down $PROG_NAME: " if [ ! -e $LOCK_FILE ]; then echo echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running." failure $"cannot stop $PROG_NAME: $PROG_NAME is not running." echo return 1; fi cd $FS_HOME $FS_FILE -stop > /dev/null 2>&1 killproc $PROG_NAME RETVAL=$? echo [ $RETVAL -eq 0 ] && rm -f $LOCK_FILE; return $RETVAL } rhstatus() { status $PROG_NAME; } case "$1" in start) start ;; stop) stop ;; status) status $PROG_NAME RETVAL=$? ;; restart) stop start ;; reload) # ;; condrestart) [ -f $PID_FILE ] && restart || : ;; *) echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" exit 1 ;; esac exit $RETVAL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/112e9099/attachment-0001.html From dynaguy at gmail.com Thu Mar 5 13:09:16 2009 From: dynaguy at gmail.com (dynaguy) Date: Thu, 5 Mar 2009 13:09:16 -0800 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot Message-ID: <2D9EE07B35234097977448BE8EDCEDA9@dell200> Hello, I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh install) and followed the instruction on the wiki installed the new Freeswitch v1.0.3. After the installation I can start FS by issus command "/usr/local/freeswitch/bin/freeswitch". After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and chmod it to 755. Reboot the server and FS is not running. If I try the start up script: [root at localhost build]# /etc/init.d/freeswitch start Starting freeswitch: [ OK ] [root at localhost build]# /etc/init.d/freeswitch status freeswitch dead but subsys locked Did I miss somthing here? Please help. Thanks a lot. dynaguy Here is the copy of the origenal freeswitch.init.centos5 ---------------------------- [root at localhost ~]# cat /etc/init.d/freeswitch #!/bin/bash # # /etc/rc.d/init.d/freeswitch # # The FreeSwitch Open Source Voice Platform # # chkconfig: 345 89 14 # description: Starts and stops the freeswitch server daemon # processname: freeswitch # config: /usr/local/freeswitch/conf/freeswitch.conf # pidfile: /usr/local/freeswitch/log/freeswitch.pid # # Source function library. . /etc/init.d/functions PROG_NAME=freeswitch PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_USER=${FS_USER-freeswitch} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} LOCK_FILE=/var/lock/subsys/freeswitch FREESWITCH_ARGS="-nc" RETVAL=0 # Source options file if [ -f /etc/sysconfig/freeswitch ]; then . /etc/sysconfig/freeswitch fi # start() { echo -n "Starting $PROG_NAME: " if [ -e $LOCK_FILE ]; then if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then echo echo -n $"$PROG_NAME is already running."; failure $"$PROG_NAME is already running."; echo return 1 fi fi cd $FS_HOME daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" echo RETVAL=$? [ $RETVAL -eq 0 ] && touch $LOCK_FILE; echo return $RETVAL } stop() { echo -n "Shutting down $PROG_NAME: " if [ ! -e $LOCK_FILE ]; then echo echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running." failure $"cannot stop $PROG_NAME: $PROG_NAME is not running." echo return 1; fi cd $FS_HOME $FS_FILE -stop > /dev/null 2>&1 killproc $PROG_NAME RETVAL=$? echo [ $RETVAL -eq 0 ] && rm -f $LOCK_FILE; return $RETVAL } rhstatus() { status $PROG_NAME; } case "$1" in start) start ;; stop) stop ;; status) status $PROG_NAME RETVAL=$? ;; restart) stop start ;; reload) # ;; condrestart) [ -f $PID_FILE ] && restart || : ;; *) echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" exit 1 ;; esac exit $RETVAL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/a48ef230/attachment-0001.html From stevecrozz at gmail.com Thu Mar 5 14:26:27 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 5 Mar 2009 14:26:27 -0800 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: References: Message-ID: <11990ade0903051426h4390602fy6f92b9020618fba8@mail.gmail.com> Your script at /etc/init.d/freeswitch is probably not referenced anywhere in your init sequence. You should read some documentation on the boot process for your system, which is probably something like this: http://www.redhat.com/docs/manuals/linux/RHL-9-Manual/ref-guide/s1-boot-init-shutdown-sysv.html Someone else might have more specifics for you. --Stephen On Thu, Mar 5, 2009 at 12:19 PM, Dyna Guy wrote: > I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh > install) and followed the instruction on the wiki installed the new > Freeswitch v1.0.3.? After the installation I can start FS by issus command > "/usr/local/freeswitch/bin/freeswitch". > > After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and > chmod it to 755. Reboot the server and FS is not running. > > If I try the start up script: > [root at localhost build]# /etc/init.d/freeswitch start > Starting freeswitch:?????????????????????????????????????? [? OK? ] > [root at localhost build]# /etc/init.d/freeswitch status > freeswitch dead but subsys locked > Did I miss somthing here? Please help. Thanks a lot. > > dynaguy > > > > > > Here is the copy of the origenal freeswitch.init.centos5 > ---------------------------- > [root at localhost ~]# cat /etc/init.d/freeswitch > #!/bin/bash > # > #?????? /etc/rc.d/init.d/freeswitch > # > #?????? The FreeSwitch Open Source Voice Platform > # > #? chkconfig: 345 89 14 > #? description: Starts and stops the freeswitch server daemon > #? processname: freeswitch > #? config: /usr/local/freeswitch/conf/freeswitch.conf > #? pidfile: /usr/local/freeswitch/log/freeswitch.pid > # > # Source function library. > . /etc/init.d/functions > PROG_NAME=freeswitch > PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} > FS_USER=${FS_USER-freeswitch} > FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} > FS_HOME=${FS_HOME-/usr/local/freeswitch} > LOCK_FILE=/var/lock/subsys/freeswitch > FREESWITCH_ARGS="-nc" > RETVAL=0 > # Source options file > if [ -f /etc/sysconfig/freeswitch ]; then > ??????? . /etc/sysconfig/freeswitch > fi > # > start() { > ??????? echo -n "Starting $PROG_NAME: " > ??????? if [ -e $LOCK_FILE ]; then > ??????????? if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then > ??????????????? echo > ??????????????? echo -n $"$PROG_NAME is already running."; > ??????????????? failure $"$PROG_NAME is already running."; > ??????????????? echo > ??????????????? return 1 > ??????????? fi > ??????? fi > ??????? cd $FS_HOME > ??????? daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE > $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" > ??????????????? echo > ??????????????? RETVAL=$? > ??????? [ $RETVAL -eq 0 ] && touch $LOCK_FILE; > ??????? echo > ??????? return $RETVAL > } > stop() { > ??????? echo -n "Shutting down $PROG_NAME: " > ??????? if [ ! -e $LOCK_FILE ]; then > ??????????? echo > ??????????? echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running." > ??????????? failure $"cannot stop $PROG_NAME: $PROG_NAME is not running." > ??????????? echo > ??????????? return 1; > ??????? fi > ??????? cd $FS_HOME > ??????? $FS_FILE -stop > /dev/null 2>&1 > ??????? killproc $PROG_NAME > ??????? RETVAL=$? > ??????? echo > ??????? [ $RETVAL -eq 0 ] &&? rm -f $LOCK_FILE; > ??????? return $RETVAL > } > rhstatus() { > ??????? status $PROG_NAME; > } > case "$1" in > ??? start) > ??????? start > ??????? ;; > ??? stop) > ??????? stop > ??????? ;; > ??? status) > ??????? status $PROG_NAME > ??????? RETVAL=$? > ??????? ;; > ??? restart) > ??????? stop > ??????? start > ??????? ;; > ??? reload) > #??????? #??????? kill -HUP or by restarting the daemons, in a manner similar > #??????? to restart above> > ??????? ;; > ??? condrestart) > ??????? [ -f $PID_FILE ] && restart || : > ??????? ;; > ??? *) > ??????? echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" > ??????? exit 1 > ??????? ;; > esac > exit $RETVAL > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From e at musinghalfwit.org Thu Mar 5 14:38:42 2009 From: e at musinghalfwit.org (Eric Liedtke) Date: Thu, 5 Mar 2009 16:38:42 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) Message-ID: <20090305223842.GA31993@pointone.com> Greetings, I've been using FS in production on this rev (I realize it's pretty far behind current) and it's been running well, save 1 issue. The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I have 2 sip profiles created , 1 per ip interface. This is being used to terminate traffic to a provider so calls are only 1 direction. They come into the private side profile, get routed via dialplan to the gateway defined in the external profile and on to the vendor. Pretty simple. I have noticed that under load (50 or so cps with ~800-900 bridged calls up) that over time some channels on the public side seem to get "stuck". Due to the nature of how this is being used , I would expect both sip profiles to show the same number of channels in use any time i do a 'sofia status' ( or at least be within a channel or 2 of each other). However after a day of heavy use I had a disparity of ~250 channels. These extra channels also seem to put some continual load on the 'system cpu' as well , reported via top. Of course due to the load on the box I have to keep logging turned way down. So I've been trying to troubleshoot it as best I can. Last night I grabbed a core file and started in with GDB today. I found the 120 or so threads that represented real active calls when I took the corefile, I also found ~250 threads that appeared to be stuck in the CS_NEW state. The backtraces on all of them looks the same, annotated below. I walked through the code path by hand , based on the bt's and I don't see how this could be happening unless it's a locking issue. But as far as I can tell each session has it's own mutex defined in the switch_core_session_t struct, so I wouldn't think they would be stepping on each other. I also would have expected if it were something of a deadlock nature it would stop processing calls all together. I grabbed the commands from the .gdbinit (super handy btw!!) and have been trolling through the variables to try to ascertain something about why these threads seem to be stuck, but am not having much luck even coming up with a scenario to try to replicate the issue. If anyone has any pointers as to where I might look next it would be greatly appreciated. We will be updating to the newest release soon, however I was hoping to nail down what is going so I can systematically replicate it and verify by testing in the lab that it is fixed , rather than just pushing the new release to produvction and hoping. Thanks in advance for any tips/pointers anyone may have. -e ......bt and bt full for a single "hung" thread #0 0xb7fd5410 in __kernel_vsyscall () #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/switch_core_state_machine.c:462 #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, obj=0x95fe270) at src/switch_core_session.c:853 #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/thread.c:138 #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 (gdb) bt full #0 0xb7fd5410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 No locals. #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/switch_core_state_machine.c:462 exception = 0 '\0' state = endstate = CS_NEW endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xb73b1720 application_state_handler = thread_id = 3085554955 env = {{__jmpbuf = {134603552, -1428248680, -1461722504, 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, 9184, 1, 2976641592, 2833244792, 3086590960, 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, 3085458203, 3086590960, 2976606624, 134564192, 2833244904}}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, obj=0x95fe270) at src/switch_core_session.c:853 session = (switch_core_session_t *) 0x95fe270 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/thread.c:138 No locals. #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 From brian at freeswitch.org Thu Mar 5 14:52:43 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 16:52:43 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <20090305223842.GA31993@pointone.com> References: <20090305223842.GA31993@pointone.com> Message-ID: Well the rules usually state that you try SVN trunk then report a jira if the problem persists but since you're 2000+ revs behind chances are we already fixed this issue. Are you using bypass media? /b On Mar 5, 2009, at 4:38 PM, Eric Liedtke wrote: > Greetings, > > I've been using FS in production on this rev (I realize it's pretty > far > behind current) and it's been running well, save 1 issue. From mrene_lists at avgs.ca Thu Mar 5 14:55:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Mar 2009 17:55:33 -0500 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <20090305223842.GA31993@pointone.com> References: <20090305223842.GA31993@pointone.com> Message-ID: <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> HI, If you suspect a bug, the place to report it is JIRA. See: http://wiki.freeswitch.org/wiki/Reporting_Bugs . This gives the whole team a way of following up on issues. Also can you upgrade to svn trunk? A lot of fixes gets committed daily, so its good to stay up to date. As you seem familiar with GDB, you may symlink the .gdbinit file in the support-d/ folder to your home directory. This will give you some FS-specific macros such as "list_sessions" which will dump a list of uuids to session pointers. In your jira, make sure you include "thread apply all bt", "list_sessions" and show channels (this one goes in FS) but PLEASE update to svn trunk and test again to see if it still happens. Also, are you using proxy/bypass media or just the default? Math On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote: > Greetings, > > I've been using FS in production on this rev (I realize it's pretty > far > behind current) and it's been running well, save 1 issue. > > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I > have > 2 sip profiles created , 1 per ip interface. This is being used to > terminate traffic to a provider so calls are only 1 direction. They > come > into the private side profile, get routed via dialplan to the gateway > defined in the external profile and on to the vendor. Pretty simple. > > I have noticed that under load (50 or so cps with ~800-900 bridged > calls up) > that over time some channels on the public side seem to get > "stuck". Due to > the nature of how this is being used , I would expect both sip > profiles to show > the same number of channels in use any time i do a 'sofia > status' ( or at least > be within a channel or 2 of each other). However after a day of > heavy use I had > a disparity of ~250 channels. These extra channels also seem to put > some > continual load on the 'system cpu' as well , reported via top. > > Of course due to the load on the box I have to keep logging turned way > down. So I've been trying to troubleshoot it as best I can. > > Last night I grabbed a core file and started in with GDB today. I > found > the 120 or so threads that represented real active calls when I took > the > corefile, I also found ~250 threads that appeared to be stuck in the > CS_NEW state. The backtraces on all of them looks the same, > annotated below. > > I walked through the code path by hand , based on the bt's and I > don't see how > this could be happening unless it's a locking issue. But as far as > I can tell > each session has it's own mutex defined in the > switch_core_session_t struct, > so I wouldn't think they would be stepping on each other. I also > would have expected > if it were something of a deadlock nature it would stop processing > calls all > together. > > I grabbed the commands from the .gdbinit (super handy btw!!) and > have been trolling > through the variables to try to ascertain something about why these > threads seem to > be stuck, but am not having much luck even coming up with a scenario > to try > to replicate the issue. > > If anyone has any pointers as to where I might look next it would be > greatly > appreciated. > > We will be updating to the newest release soon, however I was hoping > to nail down > what is going so I can systematically replicate it and verify by > testing in the lab > that it is fixed , rather than just pushing the new release to > produvction and hoping. > > Thanks in advance for any tips/pointers anyone may have. > > -e > > ......bt and bt full for a single "hung" thread > > > #0 0xb7fd5410 in __kernel_vsyscall () > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > switch_core_state_machine.c:462 > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > obj=0x95fe270) at src/switch_core_session.c:853 > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > thread.c:138 > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > (gdb) bt full > #0 0xb7fd5410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > No locals. > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > switch_core_state_machine.c:462 > exception = 0 '\0' > state = > endstate = CS_NEW > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xb73b1720 > application_state_handler = > thread_id = 3085554955 > env = {{__jmpbuf = {134603552, -1428248680, -1461722504, > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, > 9184, 1, 2976641592, 2833244792, 3086590960, > 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, > 3085458203, 3086590960, 2976606624, > 134564192, 2833244904}}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > obj=0x95fe270) at src/switch_core_session.c:853 > session = (switch_core_session_t *) 0x95fe270 > event = > event_str = 0x0 > val = > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > thread.c:138 > No locals. > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From e at musinghalfwit.org Thu Mar 5 15:19:00 2009 From: e at musinghalfwit.org (Eric Liedtke) Date: Thu, 5 Mar 2009 17:19:00 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: References: <20090305223842.GA31993@pointone.com> Message-ID: <20090305231900.GB31993@pointone.com> Yeah I know ;) I didn't open a bug because my rev was so far behind. I was just looking for any advice for where to poke next. Troubleshooting this has been a fantastic introduction to some of the inner workings of freeswitch so I was hoping to see it through and learn as I went. To answer your question no we are not using bypass media. -e It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 04:52:43PM -0600 , Brian West said: > Well the rules usually state that you try SVN trunk then report a jira > if the problem persists but since you're 2000+ revs behind chances are > we already fixed this issue. Are you using bypass media? > > /b > > On Mar 5, 2009, at 4:38 PM, Eric Liedtke wrote: > > > Greetings, > > > > I've been using FS in production on this rev (I realize it's pretty > > far > > behind current) and it's been running well, save 1 issue. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From e at musinghalfwit.org Thu Mar 5 15:22:37 2009 From: e at musinghalfwit.org (Eric Liedtke) Date: Thu, 5 Mar 2009 17:22:37 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> References: <20090305223842.GA31993@pointone.com> <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> Message-ID: <20090305232237.GC31993@pointone.com> Yup, as I mentioned to brian didn't want to clog jira with a bug that's been fixed or report against a rev 2k+ revs behind. I was trying to work through it as a learning exercise. And yeah I actually added a bunch of stuff to the list_sessions function to spit out a variety of associated variables for each session looking for a pattern somewhere to clue me into what might be happening. No proxy or bypass media here, just defaults. I will keep at it and once we update the production systems, if the problem persists I will open a bug in jira with all the neccessary goodies. Thanks -e It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM -0500 , Mathieu Rene said: > HI, > > If you suspect a bug, the place to report it is JIRA. See: http://wiki.freeswitch.org/wiki/Reporting_Bugs > . > This gives the whole team a way of following up on issues. > > Also can you upgrade to svn trunk? A lot of fixes gets committed > daily, so its good to stay up to date. > > As you seem familiar with GDB, you may symlink the .gdbinit file in > the support-d/ folder to your home directory. > This will give you some FS-specific macros such as "list_sessions" > which will dump a list of uuids to session pointers. > > In your jira, make sure you include "thread apply all bt", > "list_sessions" and show channels (this one goes in FS) but PLEASE > update to svn trunk and test again to see if it still happens. > > Also, are you using proxy/bypass media or just the default? > > Math > > On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote: > > > Greetings, > > > > I've been using FS in production on this rev (I realize it's pretty > > far > > behind current) and it's been running well, save 1 issue. > > > > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I > > have > > 2 sip profiles created , 1 per ip interface. This is being used to > > terminate traffic to a provider so calls are only 1 direction. They > > come > > into the private side profile, get routed via dialplan to the gateway > > defined in the external profile and on to the vendor. Pretty simple. > > > > I have noticed that under load (50 or so cps with ~800-900 bridged > > calls up) > > that over time some channels on the public side seem to get > > "stuck". Due to > > the nature of how this is being used , I would expect both sip > > profiles to show > > the same number of channels in use any time i do a 'sofia > > status' ( or at least > > be within a channel or 2 of each other). However after a day of > > heavy use I had > > a disparity of ~250 channels. These extra channels also seem to put > > some > > continual load on the 'system cpu' as well , reported via top. > > > > Of course due to the load on the box I have to keep logging turned way > > down. So I've been trying to troubleshoot it as best I can. > > > > Last night I grabbed a core file and started in with GDB today. I > > found > > the 120 or so threads that represented real active calls when I took > > the > > corefile, I also found ~250 threads that appeared to be stuck in the > > CS_NEW state. The backtraces on all of them looks the same, > > annotated below. > > > > I walked through the code path by hand , based on the bt's and I > > don't see how > > this could be happening unless it's a locking issue. But as far as > > I can tell > > each session has it's own mutex defined in the > > switch_core_session_t struct, > > so I wouldn't think they would be stepping on each other. I also > > would have expected > > if it were something of a deadlock nature it would stop processing > > calls all > > together. > > > > I grabbed the commands from the .gdbinit (super handy btw!!) and > > have been trolling > > through the variables to try to ascertain something about why these > > threads seem to > > be stuck, but am not having much luck even coming up with a scenario > > to try > > to replicate the issue. > > > > If anyone has any pointers as to where I might look next it would be > > greatly > > appreciated. > > > > We will be updating to the newest release soon, however I was hoping > > to nail down > > what is going so I can systematically replicate it and verify by > > testing in the lab > > that it is fixed , rather than just pushing the new release to > > produvction and hoping. > > > > Thanks in advance for any tips/pointers anyone may have. > > > > -e > > > > ......bt and bt full for a single "hung" thread > > > > > > #0 0xb7fd5410 in __kernel_vsyscall () > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > > switch_core_state_machine.c:462 > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > obj=0x95fe270) at src/switch_core_session.c:853 > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > > thread.c:138 > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > libpthread.so.0 > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > (gdb) bt full > > #0 0xb7fd5410 in __kernel_vsyscall () > > No symbol table info available. > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > No symbol table info available. > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > No symbol table info available. > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > No locals. > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > > switch_core_state_machine.c:462 > > exception = 0 '\0' > > state = > > endstate = CS_NEW > > endpoint_interface = > > driver_state_handler = (const switch_state_handler_table_t *) > > 0xb73b1720 > > application_state_handler = > > thread_id = 3085554955 > > env = {{__jmpbuf = {134603552, -1428248680, -1461722504, > > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, > > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, > > 9184, 1, 2976641592, 2833244792, 3086590960, > > 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, > > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, > > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, > > 3085458203, 3086590960, 2976606624, > > 134564192, 2833244904}}}} > > sig = > > __func__ = "switch_core_session_run" > > __PRETTY_FUNCTION__ = "switch_core_session_run" > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > obj=0x95fe270) at src/switch_core_session.c:853 > > session = (switch_core_session_t *) 0x95fe270 > > event = > > event_str = 0x0 > > val = > > __func__ = "switch_core_session_thread" > > __PRETTY_FUNCTION__ = "switch_core_session_thread" > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > > thread.c:138 > > No locals. > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > libpthread.so.0 > > No symbol table info available. > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raul at etellicom.com Thu Mar 5 15:32:52 2009 From: raul at etellicom.com (Raul Fragoso) Date: Thu, 05 Mar 2009 20:32:52 -0300 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: <2D9EE07B35234097977448BE8EDCEDA9@dell200> References: <2D9EE07B35234097977448BE8EDCEDA9@dell200> Message-ID: <1236295972.18566.38.camel@raul-laptop> Hi, and welcome to FreeSWITCH ! You've done everything right, now you only need to tell your system to run that init script during startup ;-) As root, do this: chkconfig --add freeswitch chkconfig --level 2345 freeswitch on That's all. Regards, Raul On Thu, 2009-03-05 at 13:09 -0800, dynaguy wrote: > Hello, > > I am a newbie to FS and I want learn it. So I setup a Centos 5.2 > (fresh install) and followed the instruction on the wiki installed the > new Freeswitch v1.0.3. After the installation I can start FS by issus > command "/usr/local/freeswitch/bin/freeswitch". > > > > After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch > and chmod it to 755. Reboot the server and FS is not running. > > > > If I try the start up script: > > [root at localhost build]# /etc/init.d/freeswitch start > Starting freeswitch: [ OK ] > > [root at localhost build]# /etc/init.d/freeswitch status > freeswitch dead but subsys locked > > Did I miss somthing here? Please help. Thanks a lot. > > > > dynaguy > > > > > > > > > > > > Here is the copy of the origenal freeswitch.init.centos5 > > ---------------------------- > > > [root at localhost ~]# cat /etc/init.d/freeswitch > #!/bin/bash > # > # /etc/rc.d/init.d/freeswitch > # > # The FreeSwitch Open Source Voice Platform > # > # chkconfig: 345 89 14 > # description: Starts and stops the freeswitch server daemon > # processname: freeswitch > # config: /usr/local/freeswitch/conf/freeswitch.conf > # pidfile: /usr/local/freeswitch/log/freeswitch.pid > # > > # Source function library. > . /etc/init.d/functions > > PROG_NAME=freeswitch > PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} > FS_USER=${FS_USER-freeswitch} > FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} > FS_HOME=${FS_HOME-/usr/local/freeswitch} > LOCK_FILE=/var/lock/subsys/freeswitch > FREESWITCH_ARGS="-nc" > RETVAL=0 > > # Source options file > if [ -f /etc/sysconfig/freeswitch ]; then > . /etc/sysconfig/freeswitch > fi > > # > > start() { > echo -n "Starting $PROG_NAME: " > if [ -e $LOCK_FILE ]; then > if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then > echo > echo -n $"$PROG_NAME is already running."; > failure $"$PROG_NAME is already running."; > echo > return 1 > fi > fi > cd $FS_HOME > daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE > $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" > echo > RETVAL=$? > [ $RETVAL -eq 0 ] && touch $LOCK_FILE; > echo > return $RETVAL > } > > stop() { > echo -n "Shutting down $PROG_NAME: " > if [ ! -e $LOCK_FILE ]; then > echo > echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not > running." > failure $"cannot stop $PROG_NAME: $PROG_NAME is not > running." > echo > return 1; > fi > cd $FS_HOME > $FS_FILE -stop > /dev/null 2>&1 > killproc $PROG_NAME > RETVAL=$? > echo > [ $RETVAL -eq 0 ] && rm -f $LOCK_FILE; > return $RETVAL > } > > rhstatus() { > status $PROG_NAME; > } > > case "$1" in > start) > start > ;; > stop) > stop > ;; > status) > status $PROG_NAME > RETVAL=$? > ;; > restart) > stop > start > ;; > reload) > # # kill -HUP or by restarting the daemons, in a manner similar > # to restart above> > ;; > condrestart) > [ -f $PID_FILE ] && restart || : > ;; > *) > echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" > exit 1 > ;; > esac > exit $RETVAL > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Mar 5 15:39:05 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 17:39:05 -0600 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: <1236295972.18566.38.camel@raul-laptop> References: <2D9EE07B35234097977448BE8EDCEDA9@dell200> <1236295972.18566.38.camel@raul-laptop> Message-ID: You might not wanna start it in level 2... network might not be up yet. /b On Mar 5, 2009, at 5:32 PM, Raul Fragoso wrote: > Hi, and welcome to FreeSWITCH ! > > You've done everything right, now you only need to tell your system to > run that init script during startup ;-) > As root, do this: > chkconfig --add freeswitch > chkconfig --level 2345 freeswitch on > > That's all. > > Regards, > > Raul From nik.middleton at noblesolutions.co.uk Thu Mar 5 15:39:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Mar 2009 23:39:45 -0000 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <20090305232237.GC31993@pointone.com> References: <20090305223842.GA31993@pointone.com><71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> <20090305232237.GC31993@pointone.com> Message-ID: Well if it's any consolation, I have a 4 day ish old copy of SVN and I have around 200 of these hung calls, though after an hour or so they did seem to clear. That said, FS made 138,330 call attempts today, not too shabby, and through out the call quality was as good as the first one. Not sure how to debug this one. Version: FreeSWITCH Version 1.0.trunk (12276) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric Liedtke Sent: 05 March 2009 23:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hung Channels (SVN Rev 10231) Yup, as I mentioned to brian didn't want to clog jira with a bug that's been fixed or report against a rev 2k+ revs behind. I was trying to work through it as a learning exercise. And yeah I actually added a bunch of stuff to the list_sessions function to spit out a variety of associated variables for each session looking for a pattern somewhere to clue me into what might be happening. No proxy or bypass media here, just defaults. I will keep at it and once we update the production systems, if the problem persists I will open a bug in jira with all the neccessary goodies. Thanks -e It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM -0500 , Mathieu Rene said: > HI, > > If you suspect a bug, the place to report it is JIRA. See: http://wiki.freeswitch.org/wiki/Reporting_Bugs > . > This gives the whole team a way of following up on issues. > > Also can you upgrade to svn trunk? A lot of fixes gets committed > daily, so its good to stay up to date. > > As you seem familiar with GDB, you may symlink the .gdbinit file in > the support-d/ folder to your home directory. > This will give you some FS-specific macros such as "list_sessions" > which will dump a list of uuids to session pointers. > > In your jira, make sure you include "thread apply all bt", > "list_sessions" and show channels (this one goes in FS) but PLEASE > update to svn trunk and test again to see if it still happens. > > Also, are you using proxy/bypass media or just the default? > > Math > > On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote: > > > Greetings, > > > > I've been using FS in production on this rev (I realize it's pretty > > far > > behind current) and it's been running well, save 1 issue. > > > > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I > > have > > 2 sip profiles created , 1 per ip interface. This is being used to > > terminate traffic to a provider so calls are only 1 direction. They > > come > > into the private side profile, get routed via dialplan to the gateway > > defined in the external profile and on to the vendor. Pretty simple. > > > > I have noticed that under load (50 or so cps with ~800-900 bridged > > calls up) > > that over time some channels on the public side seem to get > > "stuck". Due to > > the nature of how this is being used , I would expect both sip > > profiles to show > > the same number of channels in use any time i do a 'sofia > > status' ( or at least > > be within a channel or 2 of each other). However after a day of > > heavy use I had > > a disparity of ~250 channels. These extra channels also seem to put > > some > > continual load on the 'system cpu' as well , reported via top. > > > > Of course due to the load on the box I have to keep logging turned way > > down. So I've been trying to troubleshoot it as best I can. > > > > Last night I grabbed a core file and started in with GDB today. I > > found > > the 120 or so threads that represented real active calls when I took > > the > > corefile, I also found ~250 threads that appeared to be stuck in the > > CS_NEW state. The backtraces on all of them looks the same, > > annotated below. > > > > I walked through the code path by hand , based on the bt's and I > > don't see how > > this could be happening unless it's a locking issue. But as far as > > I can tell > > each session has it's own mutex defined in the > > switch_core_session_t struct, > > so I wouldn't think they would be stepping on each other. I also > > would have expected > > if it were something of a deadlock nature it would stop processing > > calls all > > together. > > > > I grabbed the commands from the .gdbinit (super handy btw!!) and > > have been trolling > > through the variables to try to ascertain something about why these > > threads seem to > > be stuck, but am not having much luck even coming up with a scenario > > to try > > to replicate the issue. > > > > If anyone has any pointers as to where I might look next it would be > > greatly > > appreciated. > > > > We will be updating to the newest release soon, however I was hoping > > to nail down > > what is going so I can systematically replicate it and verify by > > testing in the lab > > that it is fixed , rather than just pushing the new release to > > produvction and hoping. > > > > Thanks in advance for any tips/pointers anyone may have. > > > > -e > > > > ......bt and bt full for a single "hung" thread > > > > > > #0 0xb7fd5410 in __kernel_vsyscall () > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > > switch_core_state_machine.c:462 > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > obj=0x95fe270) at src/switch_core_session.c:853 > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > > thread.c:138 > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > libpthread.so.0 > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > (gdb) bt full > > #0 0xb7fd5410 in __kernel_vsyscall () > > No symbol table info available. > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > No symbol table info available. > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > No symbol table info available. > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > No locals. > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > > switch_core_state_machine.c:462 > > exception = 0 '\0' > > state = > > endstate = CS_NEW > > endpoint_interface = > > driver_state_handler = (const switch_state_handler_table_t *) > > 0xb73b1720 > > application_state_handler = > > thread_id = 3085554955 > > env = {{__jmpbuf = {134603552, -1428248680, -1461722504, > > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, > > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, > > 9184, 1, 2976641592, 2833244792, 3086590960, > > 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, > > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, > > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, > > 3085458203, 3086590960, 2976606624, > > 134564192, 2833244904}}}} > > sig = > > __func__ = "switch_core_session_run" > > __PRETTY_FUNCTION__ = "switch_core_session_run" > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > obj=0x95fe270) at src/switch_core_session.c:853 > > session = (switch_core_session_t *) 0x95fe270 > > event = > > event_str = 0x0 > > val = > > __func__ = "switch_core_session_thread" > > __PRETTY_FUNCTION__ = "switch_core_session_thread" > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > > thread.c:138 > > No locals. > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > libpthread.so.0 > > No symbol table info available. > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Thu Mar 5 15:43:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 17:43:35 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: References: <20090305223842.GA31993@pointone.com><71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> <20090305232237.GC31993@pointone.com> Message-ID: <063AB580-5ABD-44B8-9671-25FBE1A0483C@freeswitch.org> I would update... We fixed a few bugs related to hung calls in the past 24 hours. /b On Mar 5, 2009, at 5:39 PM, Nik Middleton wrote: > Well if it's any consolation, I have a 4 day ish old copy of SVN and I > have around 200 of these hung calls, though after an hour or so they > did > seem to clear. > > That said, FS made 138,330 call attempts today, not too shabby, and > through out the call quality was as good as the first one. Not sure > how > to debug this one. > > Version: FreeSWITCH Version 1.0.trunk (12276) From nik.middleton at noblesolutions.co.uk Thu Mar 5 15:59:44 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Mar 2009 23:59:44 -0000 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on Message-ID: Just curious here. I've always followed the fedora route but became disillusioned with the focus on the desktop rather than the server mode. Of late I've moved my servers to Centos. I felt the need for stable systems. Everyone seems to slate Centos, but to my surprise Anthony recommends Centos 5.2 which is nice to hear. Yes I know it's not bleeding edge, but I don't want that. Any reason why I should not be running Centos with FS? (I do plan on running 64 bit in future though) Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/ca34a2df/attachment.html From mszlazak at aol.com Thu Mar 5 16:35:50 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 05 Mar 2009 19:35:50 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com><49AFBFA6.45B7.0079.0@averyschools.net><49AFC537.45B7.0079.0@averyschools.net><600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org><131e01c99db9$ed440ba0$c7cc22e0$@net> <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> Message-ID: <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> If as you say "people prefer forums" then that's the nature of the target market and controlling markets can be very difficult. So you go with the market to succeed. The "build it and they will come" attitude virtually never works well. -----Original Message----- From: Kristian Kielhofner To: freeswitch-users at lists.freeswitch.org Sent: Thu, 5 Mar 2009 12:39 pm Subject: Re: [Freeswitch-users] Please end the torment A bunch of telephony geeks and a 1900 number - what could go wrong? Anyways, I too don't understand why people prefer forums. I follow dozens of mailling lists and a half a dozen e-mail addresses without ever leaving my mail client. My mail client happens to be gmail, btw: - Much more customization, filtering, etc possible than any "web forum" - "Local" copies of all messages - Search is awesome, ever hear of Google? ;) Web forums are good when you have to serve ads to people to get paid. Other than that they are certainly not the ideal tool for the job. Besides (and don't take this as an insult) - have you ever compared the web forums to the mailing lists for projects that offer both? Say what you want to say about mailing lists and IRC but the reality (usually) is the l33tz all hang out here and web forums (almost always) end up with the same groups of n00bz circling around and around trying to figure out how to accomplish even the most basic of tasks. Obviously that can go both ways but as a rule of thumb the people that are usually in a position to help others typically prefer mailing lists (probably for some of the reasons I cited above). Or maybe they are just old gray hairs too stuck in their ways. I don't know. ;) On Thu, Mar 5, 2009 at 12:43 PM, Gregory Boehnlein wrote: > You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :) > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/a0baf878/attachment-0001.html From msc at freeswitch.org Thu Mar 5 16:52:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Mar 2009 16:52:36 -0800 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: References: Message-ID: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> > Everyone seems to slate Centos, but to my surprise Anthony recommends Centos > 5.2 which is nice to hear.? Yes I know it?s not bleeding edge, but I don?t > want that. Repeat the mantra: CentOS is boring and predictable; boring and predictable is perfect for real-time telephony systems. > Any reason why I should not be running Centos with FS? (I do plan on running > 64 bit in future though) None that I can think of unless you have a super cool Linux distro that none of us have ever heard of. 64 bit OS on 64 bit hardware is a good thing. :) -MC From jason at jasonjgw.net Thu Mar 5 17:01:24 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 6 Mar 2009 12:01:24 +1100 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> Message-ID: <20090306010124.GA12699@jdc.jasonjgw.net> mszlazak at aol.com wrote: > If as you say "people prefer forums" then that's the nature of the target > market and controlling markets can be very difficult. So you go with the > market to succeed. Most free software/open-source people I've encountered prefer mailing lists and don't like being forced to use a Web interface instead (unless it's the Web interface of their preferred Web mail provider, in which case they're not being compelled to use it). For some of us, a Web forum is hard and inconvenient to use, because it substitutes the forum operator's user interface for that of the user's preferred mail client. I have reasons for choosing the mail client that I use, and if I had to work via somebody else's Web interface instead it would probably result in my not participating at all. This list can also be accessed via the Web and over NNTP at gmane.org. For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user - just connect your news reader to news.gmane.org. You can also post from the newsgroup; the first time you do so, an automated e-mail message will arrive in your inbox requesting confirmation, for spam prevention purposes. I don't know whether it is possible to post from the gmane.org Web site. They use Xapian as their search tool, which, in my experience, usually places the most relevant posts near the top of the search results. From dynaguy at gmail.com Thu Mar 5 17:07:21 2009 From: dynaguy at gmail.com (Dyna Guy) Date: Thu, 5 Mar 2009 17:07:21 -0800 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: References: <2D9EE07B35234097977448BE8EDCEDA9@dell200> <1236295972.18566.38.camel@raul-laptop> Message-ID: Thanks for all your advices. I am still struggle to make FS start. I tried few things included: chkconfig --add freeswitch chkconfig --level 345 freeswitch on I also added a user "freeswitch" The problem is : if I run "/etc/init.d./freeswitch" manually, it says [OK] like this: [root at localhost build]# /etc/init.d/freeswitch start Starting freeswitch: [ OK ] But then I did a "ps aux | grep freeswitch" it doesn't show FS running. I am not a script guru. If I run FS from commandline as root: /usr/local/freeswitch/bin/freeswitch then I can see FS running. What did I missing here? dynaguy On Thu, Mar 5, 2009 at 3:39 PM, Brian West wrote: > You might not wanna start it in level 2... network might not be up yet. > > /b > > On Mar 5, 2009, at 5:32 PM, Raul Fragoso wrote: > > > Hi, and welcome to FreeSWITCH ! > > > > You've done everything right, now you only need to tell your system to > > run that init script during startup ;-) > > As root, do this: > > chkconfig --add freeswitch > > chkconfig --level 2345 freeswitch on > > > > That's all. > > > > Regards, > > > > Raul > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/2bd53811/attachment.html From raul at etellicom.com Thu Mar 5 17:11:56 2009 From: raul at etellicom.com (Raul Fragoso) Date: Thu, 05 Mar 2009 22:11:56 -0300 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: References: <2D9EE07B35234097977448BE8EDCEDA9@dell200> <1236295972.18566.38.camel@raul-laptop> Message-ID: <1236301916.18566.39.camel@raul-laptop> Ah yes, my mistake, thanks for the correction Brian. On Thu, 2009-03-05 at 17:39 -0600, Brian West wrote: > You might not wanna start it in level 2... network might not be up yet. > > /b > > On Mar 5, 2009, at 5:32 PM, Raul Fragoso wrote: > > > Hi, and welcome to FreeSWITCH ! > > > > You've done everything right, now you only need to tell your system to > > run that init script during startup ;-) > > As root, do this: > > chkconfig --add freeswitch > > chkconfig --level 2345 freeswitch on > > > > That's all. > > > > Regards, > > > > Raul > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From davidwdan at gmail.com Thu Mar 5 17:26:03 2009 From: davidwdan at gmail.com (David Dan) Date: Thu, 5 Mar 2009 20:26:03 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <20090306010124.GA12699@jdc.jasonjgw.net> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> <20090306010124.GA12699@jdc.jasonjgw.net> Message-ID: <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> Web forums are like the wild west of the internet. They offer nothing that a mailing list and good wiki can't handle. Just go take a look at the trixbox forums. The last thing you want it for someone that is looking into freeswitch for the first time, to come across something like this (The Beginning of the End for CE), 1 click off the front page. I'd really hate to see FS go down this Mob Rule path. On Thu, Mar 5, 2009 at 8:01 PM, Jason White wrote: > mszlazak at aol.com wrote: > > If as you say "people prefer forums" then that's the nature of the target > > market and controlling markets can be very difficult. So you go with the > > market to succeed. > > Most free software/open-source people I've encountered prefer mailing lists > and don't like being forced to use a Web interface instead (unless it's the > Web interface of their preferred Web mail provider, in which case they're > not > being compelled to use it). > > For some of us, a Web forum is hard and inconvenient to use, because it > substitutes the forum operator's user interface for that of the user's > preferred mail client. I have reasons for choosing the mail client that I > use, > and if I had to work via somebody else's Web interface instead it would > probably result in my not participating at all. > > This list can also be accessed via the Web and over NNTP at gmane.org. > > For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user > - > just connect your news reader to news.gmane.org. > > You can also post from the newsgroup; the first time you do so, an > automated > e-mail message will arrive in your inbox requesting confirmation, for spam > prevention purposes. > > I don't know whether it is possible to post from the gmane.org Web site. > They > use Xapian as their search tool, which, in my experience, usually places > the > most relevant posts near the top of the search results. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/7ab00aab/attachment.html From dujinfang at gmail.com Thu Mar 5 18:00:05 2009 From: dujinfang at gmail.com (seven) Date: Fri, 6 Mar 2009 10:00:05 +0800 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49AFCD02.2000603@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> Message-ID: <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> try bridge ({effective_caller_id_name ="your_name",effective_caller_id_number="0000"}sofia/b-leg) On Mar 5, 2009, at 9:00 PM, rod wrote: > the A leg invite looks like this: > From: "Anonymous" > > it has been rewritten like this: > From: "Anonymous" > > rod > > rod wrote: >> Hi Brian, >> >> if I use the function effective_caller_id_number with my INVITE, I >> get this: >> >> From: "Anonymous" ;tag=17geyFjX5p0gS. >> >> this is not exactly what I'm looking for :p >> >> rod >> >> >> Brian West wrote: >> >>> Well this depends on how you're placing the call.. if its a standard >>> bridge you can on the A-Leg set >>> "effective_caller_id_number=000${caller_id_number}" before you call >>> bridge. >>> >>> Is the from already in the correct format? >>> >>> /b >>> >>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>> >>> >>>> Dear list, >>>> >>>> I'd like to rewrite the number in the Remote Party ID header and >>>> only in >>>> this header. >>>> >>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>> this >>>> example) in the RPID header : >>>> >>>> From: Anonymous;tag=1208367 >>>> Remote-Party-ID: >>>> >>> 123456 >>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>> >>>> should become: >>>> From: Anonymous;tag=1208367 >>>> Remote-Party-ID: >>>> >>> 000123456 >>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>> >>>> But the From field should remain unchanged. >>>> >>>> And how to strip this prefix: >>>> From: Anonymous;tag=1208367 >>>> Remote-Party-ID: >>>> >>> 000123456 >>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>> >>>> should become: >>>> From: Anonymous;tag=1208367 >>>> Remote-Party-ID: >>>> >>> 123456 >>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>> >>>> >>>> regards. >>>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Mar 5 19:59:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Mar 2009 21:59:34 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: References: <20090305223842.GA31993@pointone.com> <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> <20090305232237.GC31993@pointone.com> Message-ID: <191c3a030903051959y4a89aafaw108f41648215b35e@mail.gmail.com> if they went away by themselves they must not have been hung? On Thu, Mar 5, 2009 at 5:39 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Well if it's any consolation, I have a 4 day ish old copy of SVN and I > have around 200 of these hung calls, though after an hour or so they did > seem to clear. > > That said, FS made 138,330 call attempts today, not too shabby, and > through out the call quality was as good as the first one. Not sure how > to debug this one. > > Version: FreeSWITCH Version 1.0.trunk (12276) > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric > Liedtke > Sent: 05 March 2009 23:23 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Hung Channels (SVN Rev 10231) > > Yup, as I mentioned to brian didn't want to clog jira with a bug that's > been fixed or report against a rev 2k+ revs behind. I was trying to work > through it as a learning exercise. And yeah I actually added a bunch of > stuff to the list_sessions function to spit out a variety of associated > variables for each session looking for a pattern somewhere to clue me > into what might be happening. > > No proxy or bypass media here, just defaults. > > I will keep at it and once we update the production systems, if the > problem persists I will open a bug in jira with all the neccessary > goodies. > > Thanks > -e > > It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM > -0500 , Mathieu Rene said: > > HI, > > > > If you suspect a bug, the place to report it is JIRA. See: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > . > > This gives the whole team a way of following up on issues. > > > > Also can you upgrade to svn trunk? A lot of fixes gets committed > > daily, so its good to stay up to date. > > > > As you seem familiar with GDB, you may symlink the .gdbinit file in > > the support-d/ folder to your home directory. > > This will give you some FS-specific macros such as "list_sessions" > > which will dump a list of uuids to session pointers. > > > > In your jira, make sure you include "thread apply all bt", > > "list_sessions" and show channels (this one goes in FS) but PLEASE > > update to svn trunk and test again to see if it still happens. > > > > Also, are you using proxy/bypass media or just the default? > > > > Math > > > > On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote: > > > > > Greetings, > > > > > > I've been using FS in production on this rev (I realize it's pretty > > > > far > > > behind current) and it's been running well, save 1 issue. > > > > > > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I > > > have > > > 2 sip profiles created , 1 per ip interface. This is being used to > > > terminate traffic to a provider so calls are only 1 direction. They > > > > come > > > into the private side profile, get routed via dialplan to the > gateway > > > defined in the external profile and on to the vendor. Pretty simple. > > > > > > I have noticed that under load (50 or so cps with ~800-900 bridged > > > calls up) > > > that over time some channels on the public side seem to get > > > "stuck". Due to > > > the nature of how this is being used , I would expect both sip > > > profiles to show > > > the same number of channels in use any time i do a 'sofia > > > status' ( or at least > > > be within a channel or 2 of each other). However after a day of > > > heavy use I had > > > a disparity of ~250 channels. These extra channels also seem to put > > > > some > > > continual load on the 'system cpu' as well , reported via top. > > > > > > Of course due to the load on the box I have to keep logging turned > way > > > down. So I've been trying to troubleshoot it as best I can. > > > > > > Last night I grabbed a core file and started in with GDB today. I > > > found > > > the 120 or so threads that represented real active calls when I took > > > > the > > > corefile, I also found ~250 threads that appeared to be stuck in the > > > CS_NEW state. The backtraces on all of them looks the same, > > > annotated below. > > > > > > I walked through the code path by hand , based on the bt's and I > > > don't see how > > > this could be happening unless it's a locking issue. But as far as > > > > I can tell > > > each session has it's own mutex defined in the > > > switch_core_session_t struct, > > > so I wouldn't think they would be stepping on each other. I also > > > would have expected > > > if it were something of a deadlock nature it would stop processing > > > calls all > > > together. > > > > > > I grabbed the commands from the .gdbinit (super handy btw!!) and > > > have been trolling > > > through the variables to try to ascertain something about why these > > > > threads seem to > > > be stuck, but am not having much luck even coming up with a scenario > > > > to try > > > to replicate the issue. > > > > > > If anyone has any pointers as to where I might look next it would be > > > > greatly > > > appreciated. > > > > > > We will be updating to the newest release soon, however I was hoping > > > > to nail down > > > what is going so I can systematically replicate it and verify by > > > testing in the lab > > > that it is fixed , rather than just pushing the new release to > > > produvction and hoping. > > > > > > Thanks in advance for any tips/pointers anyone may have. > > > > > > -e > > > > > > ......bt and bt full for a single "hung" thread > > > > > > > > > #0 0xb7fd5410 in __kernel_vsyscall () > > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at > src/ > > > switch_core_state_machine.c:462 > > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > > obj=0x95fe270) at src/switch_core_session.c:853 > > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at > threadproc/unix/ > > > thread.c:138 > > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > > libpthread.so.0 > > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > (gdb) bt full > > > #0 0xb7fd5410 in __kernel_vsyscall () > > > No symbol table info available. > > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > > No symbol table info available. > > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > > No symbol table info available. > > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > > No locals. > > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at > src/ > > > switch_core_state_machine.c:462 > > > exception = 0 '\0' > > > state = > > > endstate = CS_NEW > > > endpoint_interface = > > > driver_state_handler = (const switch_state_handler_table_t *) > > > > 0xb73b1720 > > > application_state_handler = > > > thread_id = 3085554955 > > > env = {{__jmpbuf = {134603552, -1428248680, -1461722504, > > > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, > > > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, > > > 9184, 1, 2976641592, 2833244792, 3086590960, > > > 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, > > > > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, > > > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, > > > > 3085458203, 3086590960, 2976606624, > > > 134564192, 2833244904}}}} > > > sig = > > > __func__ = "switch_core_session_run" > > > __PRETTY_FUNCTION__ = "switch_core_session_run" > > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > > obj=0x95fe270) at src/switch_core_session.c:853 > > > session = (switch_core_session_t *) 0x95fe270 > > > event = > > > event_str = 0x0 > > > val = > > > __func__ = "switch_core_session_thread" > > > __PRETTY_FUNCTION__ = "switch_core_session_thread" > > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at > threadproc/unix/ > > > thread.c:138 > > > No locals. > > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > > libpthread.so.0 > > > No symbol table info available. > > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/3a3de389/attachment-0001.html From anthony.minessale at gmail.com Thu Mar 5 20:02:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Mar 2009 22:02:34 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <20090305231900.GB31993@pointone.com> References: <20090305223842.GA31993@pointone.com> <20090305231900.GB31993@pointone.com> Message-ID: <191c3a030903052002s2db4e53bi94e489a376b0ad18@mail.gmail.com> in your case you will have no choice but to update. Please do a fresh checkout as the build system has also drastically changed. On Thu, Mar 5, 2009 at 5:19 PM, Eric Liedtke wrote: > Yeah I know ;) I didn't open a bug because my rev was so far behind. I > was just looking for any advice for where to poke next. Troubleshooting > this has been a fantastic introduction to some of the inner workings of > freeswitch so I was hoping to see it through and learn as I went. > > To answer your question no we are not using bypass media. > > -e > > It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 04:52:43PM -0600 , > Brian West said: > > Well the rules usually state that you try SVN trunk then report a jira > > if the problem persists but since you're 2000+ revs behind chances are > > we already fixed this issue. Are you using bypass media? > > > > /b > > > > On Mar 5, 2009, at 4:38 PM, Eric Liedtke wrote: > > > > > Greetings, > > > > > > I've been using FS in production on this rev (I realize it's pretty > > > far > > > behind current) and it's been running well, save 1 issue. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/f9c2b14c/attachment.html From mszlazak at aol.com Thu Mar 5 22:39:50 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 06 Mar 2009 01:39:50 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com><8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com><20090306010124.GA12699@jdc.jasonjgw.net> <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> Message-ID: <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> Again, if your target market prefers lists, then go with list. If they prefer forums then it's forums. The point is that it's not about what a few like, it's about the mob but the right mob. -----Original Message----- From: David Dan To: freeswitch-users at lists.freeswitch.org Sent: Thu, 5 Mar 2009 5:26 pm Subject: Re: [Freeswitch-users] Please end the torment Web forums are like the wild west of the internet.? They offer nothing that a mailing list and good wiki can't handle. Just go take a look at the trixbox forums. The last thing you want it for someone that is looking into freeswitch for the first time, to come across something like this (The Beginning of the End for CE), 1 click off the front page.? I'd really hate to see FS go down this Mob Rule path. On Thu, Mar 5, 2009 at 8:01 PM, Jason White wrote: mszlazak at aol.com wrote: > If as you say "people prefer forums" then that's the nature of the target > market and controlling markets can be very difficult. So you go with the > market to succeed. Most free software/open-source people I've encountered prefer mailing lists and don't like being forced to use a Web interface instead (unless it's the Web interface of their preferred Web mail provider, in which case they're not being compelled to use it). For some of us, a Web forum is hard and inconvenient to use, because it substitutes the forum operator's user interface for that of the user's preferred mail client. I have reasons for choosing the mail client that I use, and if I had to work via somebody else's Web interface instead it would probably result in my not participating at all. This list can also be accessed via the Web and over NNTP at gmane.org. For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user - just connect your news reader to news.gmane.org. You can also post from the newsgroup; the first time you do so, an automated e-mail message will arrive in your inbox requesting confirmation, for spam prevention purposes. I don't know whether it is possible to post from the gmane.org Web site. They use Xapian as their search tool, which, in my experience, usually places the most relevant posts near the top of the search results. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/a29d0248/attachment.html From mashudiflexi at telkom.co.id Thu Mar 5 23:02:45 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Fri, 06 Mar 2009 14:02:45 +0700 Subject: [Freeswitch-users] make freeswitch-snapshot Message-ID: <49B0CA95.5080101@telkom.co.id> Hi Folk, i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86 here is the error : /usr/bin/ld: cannot find -lodbc collect2: ld returned 1 exit status make[2]: *** [libfreeswitch.la] Error 1 Making all in src Making all in mod making all mod_amr make[5]: *** No rule to make target `/usr/src/freeswitch-snapshot/libfreeswitch.la', needed by `mod_amr.so'. Stop. make[4]: *** [all] Error 1 make[3]: *** [mod_amr-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Did I miss something ? thank you for your support. mashudi ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From jason at jasonjgw.net Thu Mar 5 23:00:36 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 6 Mar 2009 18:00:36 +1100 Subject: [Freeswitch-users] make freeswitch-snapshot In-Reply-To: <49B0CA95.5080101@telkom.co.id> References: <49B0CA95.5080101@telkom.co.id> Message-ID: <20090306070036.GA24314@jdc.jasonjgw.net> mashudi wrote: > i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86 > here is the error : > > /usr/bin/ld: cannot find -lodbc Have you installed the ODBC library and its development headers? Are they the latest version? It's failing to find the ODBC library. From stevecrozz at gmail.com Thu Mar 5 23:03:51 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 5 Mar 2009 23:03:51 -0800 Subject: [Freeswitch-users] make freeswitch-snapshot In-Reply-To: <20090306070036.GA24314@jdc.jasonjgw.net> References: <49B0CA95.5080101@telkom.co.id> <20090306070036.GA24314@jdc.jasonjgw.net> Message-ID: <11990ade0903052303s21726d9qa956c52d52418585@mail.gmail.com> I think you need to install the debian package 'unixodbc-dev' --Stephen On Thu, Mar 5, 2009 at 11:00 PM, Jason White wrote: > mashudi wrote: >> i got error while conduct ?./make ?freeswitch-snapshot on debian 2.6 x86 >> here is the error : >> >> /usr/bin/ld: cannot find -lodbc > > Have you installed the ODBC library and its development headers? Are they the > latest version? > > It's failing to find the ODBC library. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kawarod at laposte.net Thu Mar 5 23:51:31 2009 From: kawarod at laposte.net (rod) Date: Fri, 06 Mar 2009 11:51:31 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> Message-ID: <49B0D603.502@laposte.net> using these functions like this did nothing on the SIP INVITE packet :'( seven wrote: > try > bridge > ({effective_caller_id_name > ="your_name",effective_caller_id_number="0000"}sofia/b-leg) > > On Mar 5, 2009, at 9:00 PM, rod wrote: > > >> the A leg invite looks like this: >> From: "Anonymous" >> >> it has been rewritten like this: >> From: "Anonymous" >> >> rod >> >> rod wrote: >> >>> Hi Brian, >>> >>> if I use the function effective_caller_id_number with my INVITE, I >>> get this: >>> >>> From: "Anonymous" ;tag=17geyFjX5p0gS. >>> >>> this is not exactly what I'm looking for :p >>> >>> rod >>> >>> >>> Brian West wrote: >>> >>> >>>> Well this depends on how you're placing the call.. if its a standard >>>> bridge you can on the A-Leg set >>>> "effective_caller_id_number=000${caller_id_number}" before you call >>>> bridge. >>>> >>>> Is the from already in the correct format? >>>> >>>> /b >>>> >>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>> >>>> >>>> >>>>> Dear list, >>>>> >>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>> only in >>>>> this header. >>>>> >>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>> this >>>>> example) in the RPID header : >>>>> >>>>> From: Anonymous;tag=1208367 >>>>> Remote-Party-ID: >>>>> >>>> 123456 >>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>> >>>>> should become: >>>>> From: Anonymous;tag=1208367 >>>>> Remote-Party-ID: >>>>> >>>> 000123456 >>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>> >>>>> But the From field should remain unchanged. >>>>> >>>>> And how to strip this prefix: >>>>> From: Anonymous;tag=1208367 >>>>> Remote-Party-ID: >>>>> >>>> 000123456 >>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>> >>>>> should become: >>>>> From: Anonymous;tag=1208367 >>>>> Remote-Party-ID: >>>>> >>>> 123456 >>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>> >>>>> >>>>> regards. >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From mashudiflexi at telkom.co.id Fri Mar 6 00:18:01 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Fri, 06 Mar 2009 15:18:01 +0700 Subject: [Freeswitch-users] make freeswitch-snapshot In-Reply-To: <11990ade0903052303s21726d9qa956c52d52418585@mail.gmail.com> References: <49B0CA95.5080101@telkom.co.id> <20090306070036.GA24314@jdc.jasonjgw.net> <11990ade0903052303s21726d9qa956c52d52418585@mail.gmail.com> Message-ID: <49B0DC39.1070203@telkom.co.id> Yes, it works , I would like to say thank you to Stephen Crosby & Jason White. Stephen Crosby wrote: > I think you need to install the debian package 'unixodbc-dev' > > --Stephen > > On Thu, Mar 5, 2009 at 11:00 PM, Jason White wrote: > >> mashudi wrote: >> >>> i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86 >>> here is the error : >>> >>> /usr/bin/ld: cannot find -lodbc >>> >> Have you installed the ODBC library and its development headers? Are they the >> latest version? >> >> It's failing to find the ODBC library. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From gmaruzz at celliax.org Fri Mar 6 01:39:30 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 6 Mar 2009 10:39:30 +0100 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> Message-ID: <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> On Fri, Mar 6, 2009 at 1:52 AM, Michael Collins wrote: >> Everyone seems to slate Centos, but to my surprise Anthony recommends Centos >> 5.2 which is nice to hear.? Yes I know it?s not bleeding edge, but I don?t >> want that. > > Repeat the mantra: CentOS is boring and predictable; boring and > predictable is perfect for real-time telephony systems. > >> Any reason why I should not be running Centos with FS? (I do plan on running >> 64 bit in future though) > > None that I can think of unless you have a super cool Linux distro > that none of us have ever heard of. > Maybe, but just maybe, on CentOS you can have a problem running skypiax (the skype endpoint/trunk): after a couple days of inactivity the snd-dummy ALSA driver of CentOS (at least on 32 bit) seems to go into ininterruptable sleep, causing the Skype clients to go into that state (the state seen as "D" in top). But I'm not sure about this, maybe will not be confirmed, needs more investigation. The Jira I filed for this is: http://jira.freeswitch.org/browse/MODSKYPIAX-27 I had very good overall experiences with Ubuntu 8.04 LTS Hardy, and CentOS 5.2. BTW: since roughly one month, when the sqlite assert was fixed, the build on Windows Vista seems rock solid to me. From nik.middleton at noblesolutions.co.uk Fri Mar 6 03:04:25 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Mar 2009 11:04:25 -0000 Subject: [Freeswitch-users] Setting External IP Message-ID: Hi Guys, In External.xml in sip profiles I have Can I override these for a given gateway profile? I have one gateway that's expecting a local routed IP address due to the way that it's routed, but the other one expects the public IP, hence the need to make it gateway specific Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/724761bf/attachment.html From Claudio.Cavalera at italtel.it Fri Mar 6 03:21:41 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Fri, 6 Mar 2009 12:21:41 +0100 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > Hello list, > I'm trying to track down a seg fault issue with a fs Revision: 11489 > Here is the backtrace pastebin: > http://pastebin.freeswitch.org/7009 > > but before digging the dump I would like to understand: am I the only > one having error like this in fs console: > "Error in my_thread_global_end(): 16 threads didn't exit" > > I'm asking this because googling around did not take me to > much relation > between this error and fs. > In fact as you can see the error does not have the usual fs logging > format with date time and logging level, it's just a yellow > line printed > out in console. Hello, I'm trying to track down the source of this "problem". For this reason I would like to redirect this message to a log file so that it could be compared and correlated with other logs. I'm starting fs with this command in a script: bin/freeswitch -nc -core -log /var/log/freeswitch -conf /usr/local/freeswitch/conf -db /usr/local/freeswitch/db >> /var/log/freeswitch/fs_redirection.log 2>> /var/log/freeswitch/fs_redirection.log do you think I'm safe and I will capture the error message or the -nc option could change the behaviour? Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From saeedahmad1981 at gmail.com Fri Mar 6 03:31:35 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Fri, 6 Mar 2009 12:31:35 +0100 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com><8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com><20090306010124.GA12699@jdc.jasonjgw.net><65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> Message-ID: <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> We need a poll. a) List b) Forum > (b) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/6565422d/attachment.html From helmut.kuper at ewetel.de Fri Mar 6 03:31:16 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 06 Mar 2009 12:31:16 +0100 Subject: [Freeswitch-users] Setting External IP In-Reply-To: References: Message-ID: <49B10984.8080807@ewetel.de> Hi Nik, yes you can! regards Helmut On 06.03.2009 12:04, Nik Middleton wrote: > > Hi Guys, > > > > In External.xml in sip profiles I have > > > > > > > > > > Can I override these for a given gateway profile? I have one gateway > that?s expecting a local routed IP address due to the way that it?s > routed, but the other one expects the public IP, hence the need to > make it gateway specific > > > > Regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/814b6c09/attachment.html From dujinfang at gmail.com Fri Mar 6 04:34:10 2009 From: dujinfang at gmail.com (dujinfang) Date: Fri, 6 Mar 2009 20:34:10 +0800 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B0D603.502@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> Message-ID: <46348D07-D227-42A5-A25D-A047CE5B1B63@gmail.com> How about this? bridge ({origination_caller_id_name ="your_name",origination_caller_id_number="0000"}sofia/b-leg) On Mar 6, 2009, at 3:51 PM, rod wrote: > using these functions like this did nothing on the SIP INVITE > packet :'( > > seven wrote: >> try >> bridge >> ({effective_caller_id_name >> ="your_name",effective_caller_id_number="0000"}sofia/b-leg) >> >> On Mar 5, 2009, at 9:00 PM, rod wrote: >> >> >>> the A leg invite looks like this: >>> From: "Anonymous" >>> >>> it has been rewritten like this: >>> From: "Anonymous" >>> >>> rod >>> >>> rod wrote: >>> >>>> Hi Brian, >>>> >>>> if I use the function effective_caller_id_number with my INVITE, I >>>> get this: >>>> >>>> From: "Anonymous" >>> 000000anonymous at 172.29.0.5>;tag=17geyFjX5p0gS. >>>> >>>> this is not exactly what I'm looking for :p >>>> >>>> rod >>>> >>>> >>>> Brian West wrote: >>>> >>>> >>>>> Well this depends on how you're placing the call.. if its a >>>>> standard >>>>> bridge you can on the A-Leg set >>>>> "effective_caller_id_number=000${caller_id_number}" before you >>>>> call >>>>> bridge. >>>>> >>>>> Is the from already in the correct format? >>>>> >>>>> /b >>>>> >>>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>>> >>>>> >>>>> >>>>>> Dear list, >>>>>> >>>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>>> only in >>>>>> this header. >>>>>> >>>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>>> this >>>>>> example) in the RPID header : >>>>>> >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 123456 >>>>>> @10.10.10.10 >>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> should become: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 000123456 >>>>>> @10.10.10.10 >>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> But the From field should remain unchanged. >>>>>> >>>>>> And how to strip this prefix: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 000123456 >>>>>> @10.10.10.10 >>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> should become: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 123456 >>>>>> @10.10.10.10 >>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> >>>>>> regards. >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pablosaro at gmail.com Fri Mar 6 04:54:42 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 10:54:42 -0200 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> Message-ID: <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> Hi guys, I'm using Bluewhite64 for my Linux Servers. No problems compiling and using FS, but not funny if you have dependencies problems (no yum or aptitude available to solve your problems). Really stable and secure Linux 64 bit distribution. Obviously, as the system administrator, you have to take care of keeping the system up to date with security patches and proper configurations. Regards Pablo On Fri, Mar 6, 2009 at 7:39 AM, Giovanni Maruzzelli wrote: > On Fri, Mar 6, 2009 at 1:52 AM, Michael Collins wrote: >>> Everyone seems to slate Centos, but to my surprise Anthony recommends Centos >>> 5.2 which is nice to hear.? Yes I know it?s not bleeding edge, but I don?t >>> want that. >> >> Repeat the mantra: CentOS is boring and predictable; boring and >> predictable is perfect for real-time telephony systems. >> >>> Any reason why I should not be running Centos with FS? (I do plan on running >>> 64 bit in future though) >> >> None that I can think of unless you have a super cool Linux distro >> that none of us have ever heard of. >> > > Maybe, but just maybe, on CentOS you can have a problem running > skypiax (the skype endpoint/trunk): after a couple days of inactivity > the snd-dummy ALSA driver of CentOS (at least on 32 bit) seems to go > into ininterruptable sleep, causing the Skype clients to go into that > state (the state seen as "D" in top). But I'm not sure about this, > maybe will not be confirmed, needs more investigation. The Jira I > filed for this is: http://jira.freeswitch.org/browse/MODSKYPIAX-27 > > I had very good overall experiences with Ubuntu 8.04 LTS Hardy, and > CentOS 5.2. BTW: since roughly one month, when the sqlite assert was > fixed, the build on Windows Vista seems rock solid to me. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From edpimentl at gmail.com Fri Mar 6 05:01:19 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 6 Mar 2009 08:01:19 -0500 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> Message-ID: <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> Anyone using uBuntu 8.10 and XEN? What has been your most stable VM / FS platform? Thanks in advance, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/1fa1bf9f/attachment.html From sergey.kirillov at gmail.com Fri Mar 6 05:03:08 2009 From: sergey.kirillov at gmail.com (Sergey Kirillov) Date: Fri, 06 Mar 2009 15:03:08 +0200 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card Message-ID: <49B11F0C.6040706@gmail.com> Hi everybody, I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work. Can somebody help to to configure it? Problem log (Incoming call): 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel OpenZAP/1:1/2360012 [7473c92a-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->2360012 in context default 2009-03-06 14:58:14 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer OpenZAP/1:1/2360012 to XML[1000 at default] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->1000 in context default 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/opt/freeswitch/recordings/80503820933.2009-03-06-14-58-14.wav 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:1000 at 192.168.122.1:5061;transport=udp [748a2ba2-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:1/2360012! 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:15 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:26 [NOTICE] Span:0 Q.921() I frame in invalid state ignored Here are my config files --- openzap.conf -- [span wanpipe BRI_1] name => BRI_1 trunk_type => bri b-channel => 1:1-2 d-channel => 1:3 --- openzap.conf.xml --- --- wanpipe1.conf --- [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] wp1aft1 = wanpipe1, auto, API, Comment wp1aft2 = wanpipe1, auto, API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 5 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_HW_DTMF = NO [wp1aft1] HDLC_STREAMING = NO ACTIVE_CH = 1-15.17-31 IDLE_FLAG = 0x7E MTU = 240 MRU = 240 DATA_MUX = NO TDMV_HWEC = NO [wp1aft2] HDLC_STREAMING = YES ACTIVE_CH = 16 MTU = 1500 MRU = 1500 DATA_MUX = NO TDMV_HWEC = NO From pablosaro at gmail.com Fri Mar 6 05:41:01 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 11:41:01 -0200 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> Message-ID: <247f8100903060541v10ab6605hc1d4e4de52f3db9b@mail.gmail.com> I've tried FS on ESX, but not cheap. Works good for small environments. I've never virtualized FS for big or critical systems. Pablo On Fri, Mar 6, 2009 at 11:01 AM, EdPimentl wrote: > Anyone using uBuntu 8.10 and XEN? > What has been your most stable VM / FS platform? > Thanks in advance, > -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Mar 6 06:00:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Mar 2009 08:00:28 -0600 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: References: Message-ID: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> We are considering asking CentOS to make a "FS cut" set of packages ideal for a telephony server with one install choice. On Thu, Mar 5, 2009 at 5:59 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Just curious here. > > > > I?ve always followed the fedora route but became disillusioned with the > focus on the desktop rather than the server mode. Of late I?ve moved my > servers to Centos. I felt the need for stable systems. > > > > Everyone seems to slate Centos, but to my surprise Anthony recommends > Centos 5.2 which is nice to hear. Yes I know it?s not bleeding edge, but I > don?t want that. > > > > Any reason why I should not be running Centos with FS? (I do plan on > running 64 bit in future though) > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/5e35b2f4/attachment.html From anthony.minessale at gmail.com Fri Mar 6 06:21:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Mar 2009 08:21:08 -0600 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: References: Message-ID: <191c3a030903060621r2592a89i208a7cc9cf655d91@mail.gmail.com> I recall someone asking about this before and it has to be one of the depends. Do you load anything that is not enabled by default in the standard install. It says my_ in it could it be mysql ? Let's google "my_thread_global_end()" and see... Checking http://www.google.com/search?q=my_thread_global_end() hey! http://forums.mysql.com/read.php?10,153077,153077 Not sure if its the same thing or not but it looks pretty similar. On Fri, Mar 6, 2009 at 5:21 AM, Cavalera Claudio Luigi < Claudio.Cavalera at italtel.it> wrote: > freeswitch-users-bounces at lists.freeswitch.org wrote: > > Hello list, > > I'm trying to track down a seg fault issue with a fs Revision: 11489 > > Here is the backtrace pastebin: > > http://pastebin.freeswitch.org/7009 > > > > but before digging the dump I would like to understand: am I the only > > one having error like this in fs console: > > "Error in my_thread_global_end(): 16 threads didn't exit" > > > > I'm asking this because googling around did not take me to > > much relation > > between this error and fs. > > In fact as you can see the error does not have the usual fs logging > > format with date time and logging level, it's just a yellow > > line printed > > out in console. > > > Hello, > I'm trying to track down the source of this "problem". > For this reason I would like to redirect this message to a log file so > that it could be compared and correlated with other logs. > I'm starting fs with this command in a script: > bin/freeswitch -nc -core -log /var/log/freeswitch -conf > /usr/local/freeswitch/conf -db /usr/local/freeswitch/db >> > /var/log/freeswitch/fs_redirection.log 2>> > /var/log/freeswitch/fs_redirection.log > > do you think I'm safe and I will capture the error message or the -nc > option could change the behaviour? > > Thanks, > Claudio > > > 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-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/1956e7e9/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 6 06:23:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Mar 2009 08:23:23 -0600 Subject: [Freeswitch-users] Setting External IP In-Reply-To: References: Message-ID: <191c3a030903060623p16dabe4bw5f4ed1fb2d196ca4@mail.gmail.com> gateways are children of profiles so if you need them to be separate you need to make 2 profiles and run the other one on another IP or another port. On Fri, Mar 6, 2009 at 5:04 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > In External.xml in sip profiles I have > > > > > > > > > > Can I override these for a given gateway profile? I have one gateway > that?s expecting a local routed IP address due to the way that it?s routed, > but the other one expects the public IP, hence the need to make it gateway > specific > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/faedf871/attachment.html From anthony.minessale at gmail.com Fri Mar 6 06:27:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Mar 2009 08:27:27 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> <20090306010124.GA12699@jdc.jasonjgw.net> <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> Message-ID: <191c3a030903060627y2102ebccscadf0550f7ba0d57@mail.gmail.com> Did anybody notice my email from yesterday that shows how there already is a forum on voip-info that is linked to our homepage and nobody uses it? We can't take this poll until we have a list of volunteers who would manage any new online resources. On Fri, Mar 6, 2009 at 5:31 AM, Saeed Ahmed wrote: > We need a poll. > > a) List > b) Forum > > > (b) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/0764d3c7/attachment.html From msc at freeswitch.org Fri Mar 6 08:12:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Mar 2009 08:12:37 -0800 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <191c3a030903060627y2102ebccscadf0550f7ba0d57@mail.gmail.com> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> <20090306010124.GA12699@jdc.jasonjgw.net> <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> <191c3a030903060627y2102ebccscadf0550f7ba0d57@mail.gmail.com> Message-ID: <87f2f3b90903060812s13bccf49ma7b31a1dadf5f272@mail.gmail.com> > Did anybody notice my email from yesterday that shows how there already is a > forum on voip-info that is linked to our homepage and nobody uses it? > > We can't take this poll until we have a list of volunteers who would manage > any new > online resources. Let's "end the torment of this thread" by giving an official request for the community: All of those who are willing and able to moderate a forum please stand up, go log in to voip-info.org and see what needs to be done to make it a usable forum. If you are willing to commit to managing the forum please email me at msc at freeswitch.org. There will *NOT* be a forum if we don't get at least one motivated community volunteer to take care of it. Thanks, MC From dujinfang at gmail.com Fri Mar 6 08:22:41 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 7 Mar 2009 00:22:41 +0800 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> Message-ID: We are using ubuntu 8.04 in Xen(also hosted by ubuntu 8.04, Ubuntu 8.10 is not xen friendly) as our testing server. It works well, however we only use that to test our business logic, not press test at all. On Mar 6, 2009, at 9:01 PM, EdPimentl wrote: > Anyone using uBuntu 8.10 and XEN? > What has been your most stable VM / FS platform? > Thanks in advance, > -E > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From codecomplete at free.fr Fri Mar 6 08:30:30 2009 From: codecomplete at free.fr (Fred) Date: Fri, 06 Mar 2009 17:30:30 +0100 Subject: [Freeswitch-users] Lunchbox-type PC as small server? Message-ID: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Hello I'm looking for a small, lunchbox-like PC to build a small-form factor CRM server to sell to small companies. Ideally, it should have one PCI slot so that I can stick a voice card to connect to an analog phone line and run FreeSwitch as well. I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it doesn't have room for a PCI slot, and I'm concerned about its performance. I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit pricey, and might also not be fast enough to act as a server. Are there brands/models you think I should look at? Thank you. From vikas.sharma711 at gmail.com Thu Mar 5 22:55:43 2009 From: vikas.sharma711 at gmail.com (Vikas Sharma) Date: Fri, 6 Mar 2009 12:25:43 +0530 Subject: [Freeswitch-users] About FreeSwitch Message-ID: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> HI all, I am new to this. Can anybody tell me that freeSwitch can be used as PBX indecently? Can it be integrated with other pbx as a media server? If yes, what features it has as a media server? Thnax for any help. -- vikas sharma -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/bf0aa0ce/attachment.html From rupa at rupa.com Fri Mar 6 03:36:59 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 6 Mar 2009 05:36:59 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> <20090306010124.GA12699@jdc.jasonjgw.net> <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> Message-ID: a) List On Fri, Mar 6, 2009 at 5:31 AM, Saeed Ahmed wrote: > We need a poll. > > a) List > b) Forum > > > (b) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/d76e47c7/attachment.html From chris at cloudtel.com Thu Mar 5 22:54:23 2009 From: chris at cloudtel.com (Chris Burns) Date: Thu, 5 Mar 2009 22:54:23 -0800 Subject: [Freeswitch-users] make freeswitch-snapshot In-Reply-To: <49B0CA95.5080101@telkom.co.id> References: <49B0CA95.5080101@telkom.co.id> Message-ID: <200903052254.23746.chris@cloudtel.com> apt-get install unixodbc-dev On March 5, 2009 11:02:45 pm mashudi wrote: > Hi Folk, > i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86 > here is the error : > > /usr/bin/ld: cannot find -lodbc > collect2: ld returned 1 exit status > make[2]: *** [libfreeswitch.la] Error 1 > Making all in src > Making all in mod > > making all mod_amr > make[5]: *** No rule to make target > `/usr/src/freeswitch-snapshot/libfreeswitch.la', needed by > `mod_amr.so'. Stop. > make[4]: *** [all] Error 1 > make[3]: *** [mod_amr-all] Error 1 > make[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + make install + > +----------------------------------------------+ > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Did I miss something ? > thank you for your support. > > mashudi > > > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From contactramu at gmail.com Fri Mar 6 00:51:46 2009 From: contactramu at gmail.com (Ramu) Date: Fri, 6 Mar 2009 03:51:46 -0500 Subject: [Freeswitch-users] Freeswitch and Kamailio (OpenSer) Integration Message-ID: <439e75680903060051y5021b292l76286dfe927d0337@mail.gmail.com> Hi All, I would like to setup freswitch and kamailio as follows: Kamailio acts as Proxy and Registrator Freeswitch acts as a SBC and MediaServer (voicemail) Users will be reigstered to Kamailio Kamailio forwards calls to FS to NAT FS sends back INVITE to Kamailio Kamailio will dial-out user. Bob calls Alice Bob ==INVITE ==> Kamailio ==INVITE==> FS ==INVITE==> Kamailio ==INVITE ==> Alice How can I achieve this scenario? Can you please direct me to any documentation which is available? Thanks, Ramu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/44a01947/attachment.html From nik.middleton at noblesolutions.co.uk Fri Mar 6 09:05:44 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Mar 2009 17:05:44 -0000 Subject: [Freeswitch-users] Setting External IP In-Reply-To: <191c3a030903060623p16dabe4bw5f4ed1fb2d196ca4@mail.gmail.com> References: <191c3a030903060623p16dabe4bw5f4ed1fb2d196ca4@mail.gmail.com> Message-ID: Well Here's my problem From: "FreeSWITCH" ;tag=yge6eNm7a7B0r To: CSeq: 112019702 INVITE Contact: I need to change the external IP value in the contact field to a specific IP for this gateway as I'm losing the BYE message. Is there some way of manipulating this value for a given GW? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 06 March 2009 14:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting External IP gateways are children of profiles so if you need them to be separate you need to make 2 profiles and run the other one on another IP or another port. On Fri, Mar 6, 2009 at 5:04 AM, Nik Middleton wrote: Hi Guys, In External.xml in sip profiles I have Can I override these for a given gateway profile? I have one gateway that's expecting a local routed IP address due to the way that it's routed, but the other one expects the public IP, hence the need to make it gateway specific Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/5ad5b996/attachment.html From lukasz at czerpak.eu Fri Mar 6 01:44:23 2009 From: lukasz at czerpak.eu (=?ISO-8859-2?Q?=A3ukasz_Czerpak?=) Date: Fri, 06 Mar 2009 10:44:23 +0100 Subject: [Freeswitch-users] Contact header on INVITE Message-ID: <49B0F077.6080908@czerpak.eu> Hi, The INVITE request from my FreeSWITCH contains (1.2.3.4 - my public ip): Contact: but voip provider expects: Contact: Is there any posiibility to change the Contact header value in gateway/dialplan configuration? Thanks, -- ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] From palletboy at gmail.com Fri Mar 6 09:08:30 2009 From: palletboy at gmail.com (J. G.) Date: Fri, 6 Mar 2009 12:08:30 -0500 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> Message-ID: <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> http://lmgtfy.com/?q=FreeSwitch+as+a+PBX On Fri, Mar 6, 2009 at 1:55 AM, Vikas Sharma wrote: > HI all, > I am new to this. > Can anybody tell me that freeSwitch can be used as PBX indecently? > Can it be integrated with other pbx as a media server? > If yes, what features it has as a media server? > > Thnax for any help. > > -- > vikas sharma > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/b830f297/attachment.html From red.rain.seven at gmail.com Fri Mar 6 09:10:36 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Fri, 6 Mar 2009 09:10:36 -0800 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Message-ID: <59ad9ca10903060910u179b854evfe26ccfdfc70ba90@mail.gmail.com> Check out Shuttle XPC, they have a room for video card and a PCI slot. But you have to think a about reliability when deployed in business environment. I am using this as my home server. On Fri, Mar 6, 2009 at 8:30 AM, Fred wrote: > Hello > > I'm looking for a small, lunchbox-like PC to build a small-form > factor CRM server to sell to small companies. Ideally, it should have > one PCI slot so that I can stick a voice card to connect to an analog > phone line and run FreeSwitch as well. > > I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it > doesn't have room for a PCI slot, and I'm concerned about its performance. > > I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit > pricey, and might also not be fast enough to act as a server. > > Are there brands/models you think I should look at? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/253eb70a/attachment.html From edpimentl at gmail.com Fri Mar 6 09:18:03 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 6 Mar 2009 12:18:03 -0500 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Message-ID: <9dc4a1670903060918r3aec6d53pa02f65f5140a65a9@mail.gmail.com> http://www.logicsupply.com/system_solutions http://www.advantech.com/eplatform/ -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/bcb3e442/attachment-0001.html From kristian.kielhofner at gmail.com Fri Mar 6 09:17:31 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Mar 2009 12:17:31 -0500 Subject: [Freeswitch-users] Setting External IP Message-ID: <8nRt8fFYnJQv.K5fw5rGF@smtp.gmail.com> I've never tried it but Linux routing should be able to help you with this. Create a new profile with the different address. Make sure the aliases on your network interface have a /32 netmask. Create a static route to that peer using the right outbound interface/address. Anything more than this will require iptables/iproute ip policy routing foo. Hopefully you can avoid all of that... -- Kristian Kielhofner http://blog.krisk.org -original message- Subject: [Freeswitch-users] Setting External IP From: "Nik Middleton" Date: 03/06/2009 6:05 AM Hi Guys, In External.xml in sip profiles I have Can I override these for a given gateway profile? I have one gateway that's expecting a local routed IP address due to the way that it's routed, but the other one expects the public IP, hence the need to make it gateway specific Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pablosaro at gmail.com Fri Mar 6 09:20:44 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 15:20:44 -0200 Subject: [Freeswitch-users] FS crashed Message-ID: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> Hi there, A few minutes ago, one of my FS servers crashed during normal usage (no heavy load). The hardware is Dell 1950, if you need further details please let me know. What I found on my /var/log/messages is the following: Mar 6 14:30:49 konference01 kernel: freeswitch[23798] general protection ip:7fae144e222f sp:40750540 error:0 in libc-2.7.so[7fae1446d000+14c000] I'm not familiarized with this kind of errors, but as far as I know a "general protection" occurs when a process tries to access a memory address not owned by it (probably I'm saying bullshit). The FS log does not report anything abnormal. I'm running FS svn rev 11279. Does anyone know what could happened? Thanks Pablo From nik.middleton at noblesolutions.co.uk Fri Mar 6 09:20:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Mar 2009 17:20:53 -0000 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <59ad9ca10903060910u179b854evfe26ccfdfc70ba90@mail.gmail.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> <59ad9ca10903060910u179b854evfe26ccfdfc70ba90@mail.gmail.com> Message-ID: We use the VIA mini ITX boards. Great for small offices and very stable with various fan-less options Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Henry Huang Sent: 06 March 2009 17:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Lunchbox-type PC as small server? Check out Shuttle XPC, they have a room for video card and a PCI slot. But you have to think a about reliability when deployed in business environment. I am using this as my home server. On Fri, Mar 6, 2009 at 8:30 AM, Fred wrote: Hello I'm looking for a small, lunchbox-like PC to build a small-form factor CRM server to sell to small companies. Ideally, it should have one PCI slot so that I can stick a voice card to connect to an analog phone line and run FreeSwitch as well. I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it doesn't have room for a PCI slot, and I'm concerned about its performance. I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit pricey, and might also not be fast enough to act as a server. Are there brands/models you think I should look at? Thank you. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/0099e6a6/attachment.html From gmaruzz at celliax.org Fri Mar 6 09:31:44 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 6 Mar 2009 18:31:44 +0100 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> Message-ID: <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> On Fri, Mar 6, 2009 at 6:08 PM, J. G. wrote: > http://lmgtfy.com/?q=FreeSwitch+as+a+PBX wow JG, that's pretty cool! From pablosaro at gmail.com Fri Mar 6 10:15:00 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 16:15:00 -0200 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> Message-ID: <247f8100903061015x662a1db1q78e9c0fd5a482025@mail.gmail.com> LOL On Fri, Mar 6, 2009 at 3:31 PM, Giovanni Maruzzelli wrote: > On Fri, Mar 6, 2009 at 6:08 PM, J. G. wrote: >> http://lmgtfy.com/?q=FreeSwitch+as+a+PBX > > wow JG, that's pretty cool! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Mar 6 10:19:25 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Mar 2009 12:19:25 -0600 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <247f8100903061015x662a1db1q78e9c0fd5a482025@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> <247f8100903061015x662a1db1q78e9c0fd5a482025@mail.gmail.com> Message-ID: The current default config comes out of the box as a 20 extension PBX config with various features including voicemail and conferencing. /b On Mar 6, 2009, at 12:15 PM, Pablo Hernan Saro wrote: > LOL > > On Fri, Mar 6, 2009 at 3:31 PM, Giovanni Maruzzelli > wrote: >> On Fri, Mar 6, 2009 at 6:08 PM, J. G. wrote: >>> http://lmgtfy.com/?q=FreeSwitch+as+a+PBX >> >> wow JG, that's pretty cool! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/9501950b/attachment.html From msc at freeswitch.org Fri Mar 6 10:31:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Mar 2009 10:31:23 -0800 Subject: [Freeswitch-users] FS crashed In-Reply-To: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> References: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> Message-ID: <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> > The FS log does not report anything abnormal. I'm running FS svn rev 11279. > Does anyone know what could happened? Well, you're about 1200 revs behind current SVN. A lot has been improved in the past month or two, so definitely get yourself to the latest SVN. -MC From msc at freeswitch.org Fri Mar 6 10:42:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Mar 2009 10:42:34 -0800 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> <247f8100903061015x662a1db1q78e9c0fd5a482025@mail.gmail.com> Message-ID: <87f2f3b90903061042v771c271h5c8789c2c78ff3d9@mail.gmail.com> On Fri, Mar 6, 2009 at 10:19 AM, Brian West wrote: > The current default config comes out of the box as a 20 extension PBX config > with various features including voicemail and conferencing. > /b And, unlike many other systems, you don't have to pay to increase the number of extensions on the system. In conf/directory/default/ there are 20 xml files: 1000.xml ... 1019.xml You can create new extensions by creating new xml files here. Then locate the "Local_Extension" in conf/dialplan/default.xml and modify the regex, which by default is this: If you want 1000~1299 to be local extensions then do this: too easy... -MC From woof at nortel.com Fri Mar 6 11:09:13 2009 From: woof at nortel.com (Andy Spitzer) Date: Fri, 06 Mar 2009 14:09:13 -0500 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> Message-ID: Woof! On Fri, 06 Mar 2009 01:55:43 -0500, Vikas Sharma wrote: > Can it be integrated with other pbx as a media server? Yep, it sure can: http://sipx-wiki.calivia.com/index.php/SipXivr --Woof! From pablosaro at gmail.com Fri Mar 6 11:37:03 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 17:37:03 -0200 Subject: [Freeswitch-users] FS crashed In-Reply-To: <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> References: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> Message-ID: <247f8100903061137m405fa3act93aa2875a8136df4@mail.gmail.com> Thanks for your answer Michael. What do you recommend for production environments: the latest SVN or 1.0.3 tarball? Pablo On Fri, Mar 6, 2009 at 4:31 PM, Michael Collins wrote: >> The FS log does not report anything abnormal. I'm running FS svn rev 11279. >> Does anyone know what could happened? > > Well, you're about 1200 revs behind current SVN. A lot has been > improved in the past month or two, so definitely get yourself to the > latest SVN. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Mar 6 11:40:49 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Mar 2009 13:40:49 -0600 Subject: [Freeswitch-users] FS crashed In-Reply-To: <247f8100903061137m405fa3act93aa2875a8136df4@mail.gmail.com> References: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> <247f8100903061137m405fa3act93aa2875a8136df4@mail.gmail.com> Message-ID: latest SVN is what I would give a try. /b On Mar 6, 2009, at 1:37 PM, Pablo Hernan Saro wrote: > Thanks for your answer Michael. > What do you recommend for production environments: the latest SVN or > 1.0.3 tarball? > > Pablo From pablosaro at gmail.com Fri Mar 6 11:44:05 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 17:44:05 -0200 Subject: [Freeswitch-users] FS crashed In-Reply-To: References: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> <247f8100903061137m405fa3act93aa2875a8136df4@mail.gmail.com> Message-ID: <247f8100903061144q804d848i8476a5168860b8a2@mail.gmail.com> thx Brian. On Fri, Mar 6, 2009 at 5:40 PM, Brian West wrote: > latest ?SVN is what I would give a try. > > /b > > On Mar 6, 2009, at 1:37 PM, Pablo Hernan Saro wrote: > >> Thanks for your answer Michael. >> What do you recommend for production environments: the latest SVN or >> 1.0.3 tarball? >> >> Pablo > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From hads at nice.net.nz Fri Mar 6 11:57:34 2009 From: hads at nice.net.nz (Hadley Rich) Date: Sat, 7 Mar 2009 08:57:34 +1300 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Message-ID: <200903070857.34614.hads@nice.net.nz> On Saturday 07 March 2009 05:30:30 Fred wrote: > I'm looking for a small, lunchbox-like PC to build a small-form > factor CRM server to sell to small companies. Ideally, it should have > one PCI slot so that I can stick a voice card to connect to an analog > phone line and run FreeSwitch as well. > > I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it > doesn't have room for a PCI slot, and I'm concerned about its performance. > > I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit > pricey, and might also not be fast enough to act as a server. > > Are there brands/models you think I should look at? The Intel range of ATOM Mini-ITX boards. http://www.intel.com/products/desktop/motherboards/D945GCLF/D945GCLF- overview.htm hads -- http://nicegear.co.nz New Zealands Open Source Hardware Supplier From kristian.kielhofner at gmail.com Fri Mar 6 14:01:56 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Mar 2009 17:01:56 -0500 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> References: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> Message-ID: <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> Why not just provide a kickstart file on freeswitch.org? It's pretty easy to pass them to the installer over the network and/or add them onto existing ISOs and other bootable media... I'd be happy to write/maintain such a thing. After all, I use them myself! ;) On Fri, Mar 6, 2009 at 9:00 AM, Anthony Minessale wrote: > We are considering asking CentOS to make a "FS cut" set of packages ideal > for a telephony server with one install choice. > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From stevecrozz at gmail.com Fri Mar 6 14:07:55 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Fri, 6 Mar 2009 14:07:55 -0800 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> References: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> Message-ID: <11990ade0903061407t1b262df5uce7aa1bf53015143@mail.gmail.com> I wasn't going to say anything, but since somebody already mentioned ubuntu, I'll add that I'm using Hardy Heron LTS as well. I haven't had a single issue with that, I think it was a great choice. --Stephen On Fri, Mar 6, 2009 at 2:01 PM, Kristian Kielhofner wrote: > Why not just provide a kickstart file on freeswitch.org? ?It's pretty > easy to pass them to the installer over the network and/or add them > onto existing ISOs and other bootable media... > > I'd be happy to write/maintain such a thing. ?After all, I use them myself! ;) > > On Fri, Mar 6, 2009 at 9:00 AM, Anthony Minessale > wrote: >> We are considering asking CentOS to make a "FS cut" set of packages ideal >> for a telephony server with one install choice. >> > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Mar 6 14:28:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Mar 2009 14:28:23 -0800 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> References: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> Message-ID: <87f2f3b90903061428k30028bb9ldca1ccb3349907e0@mail.gmail.com> On Fri, Mar 6, 2009 at 2:01 PM, Kristian Kielhofner wrote: > Why not just provide a kickstart file on freeswitch.org? ?It's pretty > easy to pass them to the installer over the network and/or add them > onto existing ISOs and other bootable media... > > I'd be happy to write/maintain such a thing. ?After all, I use them myself! ;) You're hired! :) -MC From gcd at i.ph Fri Mar 6 15:03:41 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sat, 7 Mar 2009 07:03:41 +0800 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Message-ID: <7d0bfd8c0903061503s27df5706q6671fbec46bd3dc6@mail.gmail.com> take a look at intel atom mobo d945gclf2 (dual-core). it has one PCI slot and an S-Video and VGA video ports. rhino is using this platform. On Sat, Mar 7, 2009 at 12:30 AM, Fred wrote: > Hello > > I'm looking for a small, lunchbox-like PC to build a small-form > factor CRM server to sell to small companies. Ideally, it should have > one PCI slot so that I can stick a voice card to connect to an analog > phone line and run FreeSwitch as well. > > I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it > doesn't have room for a PCI slot, and I'm concerned about its performance. > > I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit > pricey, and might also not be fast enough to act as a server. > > Are there brands/models you think I should look at? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/1b7cc5e6/attachment.html From edpimentl at gmail.com Fri Mar 6 16:20:42 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 6 Mar 2009 19:20:42 -0500 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7d0bfd8c0903061503s27df5706q6671fbec46bd3dc6@mail.gmail.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> <7d0bfd8c0903061503s27df5706q6671fbec46bd3dc6@mail.gmail.com> Message-ID: <9dc4a1670903061620n443fe2c5g6e26296ce1cb33a0@mail.gmail.com> If using Intel D945..... MoBo.. then check this out http://www.mini-box.com/Intel-D945GCLF2-Mini-ITX-Motherboard http://www.mini-box.com/M350-universal-mini-itx-enclosure;jsessionid=0a0101421f43385f65a16e0c426987a8e1ec3d627373.e3eSc3iSaN0Le34Pa38Ta38Pahf0 -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/eceac58c/attachment.html From casteven at gmail.com Fri Mar 6 17:16:36 2009 From: casteven at gmail.com (Campbell Steven) Date: Sat, 07 Mar 2009 14:16:36 +1300 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <9dc4a1670903061620n443fe2c5g6e26296ce1cb33a0@mail.gmail.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> <7d0bfd8c0903061503s27df5706q6671fbec46bd3dc6@mail.gmail.com> <9dc4a1670903061620n443fe2c5g6e26296ce1cb33a0@mail.gmail.com> Message-ID: <1236388596.19995.6819.camel@macmini> I second the advantech boxes someone mentioned further up the thread, we have a few of the ARK-3000 series out there and they work great. http://www.advantech.com/products/Four-LAN-Ports-Compact-System/mod_1-2JKEGT.aspx Campbell On Fri, 2009-03-06 at 19:20 -0500, EdPimentl wrote: > If using Intel D945..... MoBo.. then check this out > http://www.mini-box.com/Intel-D945GCLF2-Mini-ITX-Motherboard > http://www.mini-box.com/M350-universal-mini-itx-enclosure;jsessionid=0a0101421f43385f65a16e0c426987a8e1ec3d627373.e3eSc3iSaN0Le34Pa38Ta38Pahf0 > > -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/9814faf5/attachment-0001.html From jaybinks at gmail.com Fri Mar 6 18:41:52 2009 From: jaybinks at gmail.com (jay binks) Date: Sat, 7 Mar 2009 12:41:52 +1000 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <11990ade0903061407t1b262df5uce7aa1bf53015143@mail.gmail.com> References: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> <11990ade0903061407t1b262df5uce7aa1bf53015143@mail.gmail.com> Message-ID: I personally like and use Debian .. all my boxes are debian 4... havnt looked at using debian 5 yet. Jay On Sat, Mar 7, 2009 at 8:07 AM, Stephen Crosby wrote: > I wasn't going to say anything, but since somebody already mentioned > ubuntu, I'll add that I'm using Hardy Heron LTS as well. I haven't had > a single issue with that, I think it was a great choice. > > --Stephen > > On Fri, Mar 6, 2009 at 2:01 PM, Kristian Kielhofner > wrote: > > Why not just provide a kickstart file on freeswitch.org? It's pretty > > easy to pass them to the installer over the network and/or add them > > onto existing ISOs and other bootable media... > > > > I'd be happy to write/maintain such a thing. After all, I use them > myself! ;) > > > > On Fri, Mar 6, 2009 at 9:00 AM, Anthony Minessale > > wrote: > >> We are considering asking CentOS to make a "FS cut" set of packages > ideal > >> for a telephony server with one install choice. > >> > > > > > > -- > > Kristian Kielhofner > > http://blog.krisk.org > > http://www.submityoursip.com > > http://www.astlinux.org > > http://www.star2star.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/fbc6d0b7/attachment.html From kjv at ken-ton.com Sat Mar 7 06:10:25 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Sat, 7 Mar 2009 09:10:25 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <49AFBFA6.45B7.0079.0@averyschools.net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: <114ACB3A-3637-4FEA-A539-165A34DEAD09@ken-ton.com> Ben; > Can you imagine me injecting a question about a SIP profile into > that conversation?? Don't be bashful, just jump right in: > If I'm going into #freeswitch at 11pm at night, it's probably > because I really need some help with some problem I've run into > after hours. Can you imagine me injecting a question about a SIP > profile into that conversation?? > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though [23:12] Excuse me, I hate to interrupt, but I have a FreeSwitch question regarding (foo), can anyone help me with that? Point being that you're certainly not going to get any assistance if you neglect to ask for it. I'm not sure I speak for all of us, but we volunteer our time in the IRC channel, and on the Wiki. Since we're not paid to give our advice and/or professional opinions, we're going to "goof off" when there's no one there asking for help. We're also going to goof off when there's people in there that need help, but are too bashful or afraid to ask for it. It's human nature, and it will happen. It even happens in the work place. Should you doubt me, just watch around the water cooler at work. Guaranteed you'll see a couple folks engaged in conversation that is non-work-related. You'd probably have gotten your answer a lot sooner by asking for it in IRC as outlined above, rather that too whine about the conversations of the people that volunteer their time to help you. Summary: If you need help, ask. If you don't ask, then the only outcome of that lack of action is what? I could say more, a lot more, but I think I've made my point. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-448-3009 x0 On Mar 5, 2009, at 12:03 PM, Ben Holtsclaw wrote: > I agree with Harry. I do not like the mailing list. Those that do > like the mailing list always advocate Nabble. For those that > advocate that solution, do you even realize that you can't post on > Nabble unless you are subscribed to the mailing list? I am also not > a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in > years! I think I uninstalled my IRC client at the same time I > uninstalled my NNTP reader. Most of the time, I actually find it > difficult to obtain support in the #freeswitch channel. Once you ask > the question, if somebody doesn't happen to be there that knows the > answer, then you're screwed. How many times have I asked a question > only to wait 30 seconds and then see, "anthm has joined > #freeswitch." Crap...do I ask the question again? I have found the > conversation in #openzap to be much more focused. Thank goodness I'm > using that module! In that channel, I never see conversations about > cd burners, somebody's girlfriend in South America, or off color > jokes about someone's sexual proclivity. And because I know I'll get > flamed for saying that, just look at this: > > [23:10] <{tasker}> me, too, but i'm a different animal > [23:10] <{tasker}> in NY and in Miami i went nutz > [23:10] lol > [23:10] * jefferai is now known as lollerai > [23:10] yeah i love her > [23:10] <{tasker}> latinas everywhere > [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left > #freeswitch > [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection > timed out)) > [23:11] here its blond blue eyed girls > [23:11] * lollerai is now known as lolferai > [23:11] brazilians... hopefully she's hot. i've seen some > pretty dodgy looking chicks from there > [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, > "yeaaaaaaaaah, it's just for a few days" > [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though > > If I'm going into #freeswitch at 11pm at night, it's probably > because I really need some help with some problem I've run into > after hours. Can you imagine me injecting a question about a SIP > profile into that conversation?? > > ALL that aside... I'm willing to use a carrier pigeon if that's the > way the three primary developers wish to communicate. They have been > instrumental in getting my project where it is today. You know the > saying... beggars can't be choosers. > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > >>> On 3/5/2009 at 5:14 AM, "J Mann/Harry" > wrote: > No, I've yet to contribute anything, I barely have my system doing > what I want. But I REALLY love Freeswitch and I want to see it BURY > Asterisk. (Windows server user here) > > I've been struggling with the XML configs, trying to figure out what > does what and where! That's fine, I'm used to it. What I'm NOT used to > is the total lack of a forum-based community to join and participate > in! Where can users SHARE their configs, help each other, learn from > each others mistakes? No DEV forum? I'm speechless. > > Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" > at best! I want stickies, a forum for noobs, converts, a dev forum... > So on... > > "WELCOME To FreeSWITCH!" > > Am I asking too much here? A FORUM? > > I can't see how you can spark interest when we're so sorely lacking > the most basic and widely used community environments on the net! > > HELP SOMEBODY? Install SMF ASAP! > http://www.simplemachines.org > > BTW: I hate people who voice problems without offering viable > solutions in the process... Disgusting! So if someone can offer up a > simple hosting account, Control Panel 10, Windows Linux whatever... > I'll be more than happy to have SMF setup and receiving user > registrations within 24 hours! I've done it dozens of times before. I > will then gladly turn over the keys to the kingdom to the "powers that > be" and take a backseat, being simply a happy user from that point on! > > Please folks, please. I'm dying over here and I'm sure I'm not alone! > I'm searching Google and finding nothing!! FORUMS! > > Harry (email me here) > switchserver at gmail.com (my FS email) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/57b95830/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/57b95830/attachment.bin From dave at 3c.co.uk Fri Mar 6 06:25:59 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 06 Mar 2009 07:25:59 -0700 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B0D603.502@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> Message-ID: <49B13277.9090505@3c.co.uk> Hi Rod, You can set it directly: ;screen=yes;privacy=off]]> --Dave > using these functions like this did nothing on the SIP INVITE packet :'( > > seven wrote: > >> try >> bridge >> ({effective_caller_id_name >> ="your_name",effective_caller_id_number="0000"}sofia/b-leg) >> >> On Mar 5, 2009, at 9:00 PM, rod wrote: >> >> >> >>> the A leg invite looks like this: >>> From: "Anonymous" >>> >>> it has been rewritten like this: >>> From: "Anonymous" >>> >>> rod >>> >>> rod wrote: >>> >>> >>>> Hi Brian, >>>> >>>> if I use the function effective_caller_id_number with my INVITE, I >>>> get this: >>>> >>>> From: "Anonymous" ;tag=17geyFjX5p0gS. >>>> >>>> this is not exactly what I'm looking for :p >>>> >>>> rod >>>> >>>> >>>> Brian West wrote: >>>> >>>> >>>> >>>>> Well this depends on how you're placing the call.. if its a standard >>>>> bridge you can on the A-Leg set >>>>> "effective_caller_id_number=000${caller_id_number}" before you call >>>>> bridge. >>>>> >>>>> Is the from already in the correct format? >>>>> >>>>> /b >>>>> >>>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> Dear list, >>>>>> >>>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>>> only in >>>>>> this header. >>>>>> >>>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>>> this >>>>>> example) in the RPID header : >>>>>> >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 123456 >>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> should become: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 000123456 >>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> But the From field should remain unchanged. >>>>>> >>>>>> And how to strip this prefix: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 000123456 >>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> should become: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 123456 >>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> >>>>>> regards. >>>>>> >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/036723ce/attachment-0001.html From anthony.minessale at gmail.com Sat Mar 7 08:05:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 7 Mar 2009 10:05:37 -0600 Subject: [Freeswitch-users] Contact header on INVITE In-Reply-To: <49B0F077.6080908@czerpak.eu> References: <49B0F077.6080908@czerpak.eu> Message-ID: <191c3a030903070805y1a8bdcf6j4be5f0c165f907af@mail.gmail.com> I added the param extension-in-contact (true/false) to latest trunk. Be aware it's broken behavior for your provider to demand a certain username in the contact. The reason we put gw+ in the contact user is it's the only guaranteed way to find out what gateway the call was associated with. We can of course put it in the params on the contact but we know many sip endpoints eat off all the params from your contact when you register. So we use the contact user to avoid this issue. If you use this new parameter and set it to true, the contact will be whatever is in the "extension" parameter which defaults to the same as the username, so just setting this parameter should do what you want but it will also disable our ability to associate the call with the gateway it was placed from. 2009/3/6 ?ukasz Czerpak > Hi, > > The INVITE request from my FreeSWITCH contains (1.2.3.4 - my public ip): > > Contact: > > but voip provider expects: > > Contact: > > > Is there any posiibility to change the Contact header value in > gateway/dialplan configuration? > > > Thanks, > > -- > ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/875723ea/attachment.html From nik.middleton at noblesolutions.co.uk Sat Mar 7 15:37:23 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 7 Mar 2009 23:37:23 -0000 Subject: [Freeswitch-users] Getting a sip trace on the console Message-ID: Hi Guys, I'm trying to debug some SIP messaging issues. Is there a way of doing the Asterisk equivalent of SIP Debug so I can see what's being sent? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/cc90119a/attachment.html From jason at jasonjgw.net Sat Mar 7 15:56:49 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 8 Mar 2009 10:56:49 +1100 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: Message-ID: <20090307235649.GA10047@jdc.jasonjgw.net> Nik Middleton wrote: > I'm trying to debug some SIP messaging issues. Is there a way of doing > the Asterisk equivalent of SIP Debug so I can see what's being sent? http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP You can also use the sofia loglevel command from fs_cli, e.g., sofia loglevel 9 which will give you everything but a SIP trace. For the SIP trace, you'll need to set the environment variables as documented on the above page, or turn on tracing in the SIP profile configuration. From hads at nice.net.nz Sat Mar 7 16:00:15 2009 From: hads at nice.net.nz (Hadley Rich) Date: Sun, 8 Mar 2009 13:00:15 +1300 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: Message-ID: <200903081300.15833.hads@nice.net.nz> On Sunday 08 March 2009 12:37:23 Nik Middleton wrote: > I'm trying to debug some SIP messaging issues. Is there a way of doing > the Asterisk equivalent of SIP Debug so I can see what's being sent? TPORT_LOG=1 ./freeswitch will do it for you. http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP hads -- http://nicegear.co.nz New Zealands Open Source Hardware Supplier From rdmitry0911 at yandex.ru Sat Mar 7 12:12:28 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Sat, 7 Mar 2009 12:12:28 -0800 (PST) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <22339162.post@talk.nabble.com> References: <22339162.post@talk.nabble.com> Message-ID: <22391395.post@talk.nabble.com> Hi, I found one more interesting thing related to this problem. It seems, that skype protocol signaling depends on skype name used in skypiax. I was able to find couple of names, which work fine. However if I copy all settings from these lucky skype profiles to other profiles, which don't work, they do not become working anyway. In any event, I think "TRANSFERRED" and "TRANSFERRING" are valid keywords in skype signaling protocol and need to be included in skypiax_signaling_read() function Best Regards, Dmitry -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22391395.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sat Mar 7 16:54:56 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Mar 2009 18:54:56 -0600 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: <200903081300.15833.hads@nice.net.nz> References: <200903081300.15833.hads@nice.net.nz> Message-ID: if you have latest SVN you can do this: sofia profile XXX siptrace on /b On Mar 7, 2009, at 6:00 PM, Hadley Rich wrote: > On Sunday 08 March 2009 12:37:23 Nik Middleton wrote: >> I'm trying to debug some SIP messaging issues. Is there a way of >> doing >> the Asterisk equivalent of SIP Debug so I can see what's being sent? > > TPORT_LOG=1 ./freeswitch will do it for you. > > http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP > > hads -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/ad842ac7/attachment.html From brian at freeswitch.org Sat Mar 7 16:55:53 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Mar 2009 18:55:53 -0600 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: <200903081300.15833.hads@nice.net.nz> References: <200903081300.15833.hads@nice.net.nz> Message-ID: <4070E206-0B7D-473E-97FA-4401F220968F@freeswitch.org> Also sofia loglevel [0-9] No need to restart FS to turn those on or off. ;) again Latest SVN. /b On Mar 7, 2009, at 6:00 PM, Hadley Rich wrote: > http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/04670f7f/attachment.html From mrene_lists at avgs.ca Sat Mar 7 19:08:03 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 7 Mar 2009 22:08:03 -0500 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: <4070E206-0B7D-473E-97FA-4401F220968F@freeswitch.org> References: <200903081300.15833.hads@nice.net.nz> <4070E206-0B7D-473E-97FA-4401F220968F@freeswitch.org> Message-ID: <1E8D1AD8-02BF-4D53-95C4-014F47E94E6A@avgs.ca> only sofia profile [name] siptrace [on/off] is required to get the messages on the console sofia loglevel is used to get debugging logs from sofia's internals. (the sip library) Math On 7-Mar-09, at 7:55 PM, Brian West wrote: > Also > > sofia loglevel nth_server|nua|soa|sresolv|stun> [0-9] > > No need to restart FS to turn those on or off. ;) again Latest SVN. > > /b > > > On Mar 7, 2009, at 6:00 PM, Hadley Rich wrote: > >> http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/b77a55bb/attachment-0001.html From lfurrea at gmail.com Sat Mar 7 20:40:31 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Sat, 7 Mar 2009 22:40:31 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <114ACB3A-3637-4FEA-A539-165A34DEAD09@ken-ton.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <114ACB3A-3637-4FEA-A539-165A34DEAD09@ken-ton.com> Message-ID: Same thing happened to me in regards IRC, I had not used it for years before getting into FS, but as a total newbie I can say that IRC is really good to get things going quickly and all those No rocket science questions we have. I have had people helping realtime looking at pastebin's and stuff, which is a really good thing. When one needs to elaborate on a question maybe mailing list/forum fits better, but what I know in my experience is that eventually you will get the answer thanks to the efforts of the community with the current set of tools. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/fe5d075f/attachment.html From jason at jasonjgw.net Sat Mar 7 22:00:36 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 8 Mar 2009 17:00:36 +1100 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: <200903081300.15833.hads@nice.net.nz> Message-ID: <20090308060036.GA19133@jdc.jasonjgw.net> Brian West wrote: > if you have latest SVN you can do this: > > sofia profile XXX siptrace on Thanks. That was enough to prompt me to recompile with rev. 12516. From diego.viola at gmail.com Sat Mar 7 21:41:20 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 8 Mar 2009 02:41:20 -0300 Subject: [Freeswitch-users] mod_nibblebill question: DB Error while updating cash! Message-ID: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> Hello, I'm trying to give mod_nibblebill a try, I compiled it and created the DB, set up ODBC, etc. I'm using MySQL. This is how I created the db: CREATE TABLE accounts ( id int NOT NULL PRIMARY KEY, name VARCHAR(255), cash double precision NOT NULL ); However when I try to make a call I get this: 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error while updating cash! I have this also on my user directory: Any ideas? Thanks. From diego.viola at gmail.com Sat Mar 7 22:28:42 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 8 Mar 2009 03:28:42 -0300 Subject: [Freeswitch-users] mod_nibblebill question: DB Error while updating cash! In-Reply-To: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> References: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> Message-ID: <86a32abc0903072228x67939d14x81a0c5e692cf67ac@mail.gmail.com> Oh, I noticed the billing actually works, it discounts from my credit but I still get that message, even if the update works. "2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error while updating cash!" Thanks. On Sun, Mar 8, 2009 at 3:41 AM, Diego Viola wrote: > Hello, > > I'm trying to give mod_nibblebill a try, I compiled it and created the > DB, set up ODBC, etc. I'm using MySQL. > > This is how I created the db: > > ?CREATE TABLE accounts > ?( > ? id int NOT NULL PRIMARY KEY, > ? name VARCHAR(255), > ? cash double precision NOT NULL > ?); > > However when I try to make a call I get this: > > 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error > while updating cash! > > I have this also on my user directory: > > ? ? ? > ? ? ? > > > Any ideas? > > Thanks. > From rdmitry0911 at yandex.ru Sun Mar 8 03:17:49 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Sun, 8 Mar 2009 03:17:49 -0700 (PDT) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name Message-ID: <22339162.post@talk.nabble.com> Hi all, I've got a strange problem with skypiax. I successfully installed freeswitch revision 12408 with skypiax and configured 2 skype channels with different names. When I try to call both names one by one or simultaneously, everything goes fine. But when I try to place a second call to the same skype name which is busy with the first call, I get the following message: 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ][skypiax1 ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized and second call can't get thru. I can hear call progress tones only. After about 5 seconds the message ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ][skypiax1 ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized occurs and I can hear only silence after that. Does anybody know what might cause such a problem? I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) Any help would be very much appreciated. Best regards, Dmitry -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Sun Mar 8 05:11:29 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Mar 2009 12:11:29 -0000 Subject: [Freeswitch-users] Freeswitch IAX support Message-ID: Hi Guys, Now that IAX has been published as an RFC (http://www.rfc-editor.org/authors/rfc5456.txt) are there any plans to support registrations? Not a moan, just curious as to the road map. A lot of my users have Asterisk PBX's using IAX and I'd love to replace my Asterisk central server with FS to better serve them. Yes I know I could get them to move to using SIP, but there's a lot of them. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090308/f00e8a82/attachment.html From Prometheus001 at gmx.net Sun Mar 8 05:57:45 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 08 Mar 2009 13:57:45 +0100 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: Message-ID: <49B3C0C9.7040901@gmx.net> I use the ngrep tool on the OS console and write the output to a file: ngrep -d any port 5060 -W byline >outfile.txt Best regards Peter Nik Middleton schrieb: > > Hi Guys, > > > > I?m trying to debug some SIP messaging issues. Is there a way of > doing the Asterisk equivalent of SIP Debug so I can see what?s being sent? > > > > Regards, > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Sun Mar 8 07:00:23 2009 From: dujinfang at gmail.com (dujinfang) Date: Sun, 8 Mar 2009 22:00:23 +0800 Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <22339162.post@talk.nabble.com> References: <22339162.post@talk.nabble.com> Message-ID: I only use skypiax do outbound calls by using the ANY interface, and it works pretty well. It will be really cool if multi channels can call in to a single account. However, AFAIK, the skype client on my computer, if the second call come in the first channel will change to "hold". It does can accept other incoming calls, but only one channel can be active. I haven't play it into deep and not sure if one skype instance can transfer to another account or another instance on busy. On Mar 8, 2009, at 6:17 PM, rdmitry wrote: > > Hi all, > > I've got a strange problem with skypiax. I successfully installed > freeswitch > revision 12408 with skypiax and configured 2 skype channels with > different > names. When I try to call both names one by one or simultaneously, > everything goes fine. But when I try to place a second call to the > same > skype name which is busy with the first call, I get the following > message: > > 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 > skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ] > [skypiax1 > ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized > > and second call can't get thru. I can hear call progress tones only. > After > about 5 seconds the message > > ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 > skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ] > [skypiax1 > ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized > > occurs and I can hear only silence after that. > > Does anybody know what might cause such a problem? > > I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) > > Any help would be very much appreciated. > > Best regards, Dmitry > > > > > > -- > View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rdmitry0911 at yandex.ru Sun Mar 8 07:20:54 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Sun, 8 Mar 2009 07:20:54 -0700 (PDT) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: References: <22339162.post@talk.nabble.com> Message-ID: <22398443.post@talk.nabble.com> Yes, it can. And as I can see it, freeswitch always gets control over second call, if it finds a free skypiax channel. The problem is that for some skype names it happens as skypiax expects it shoud happen, and for other names it happens the way skypiax doesn't expect and considers it as an exeption. For example, skype names like test-lineX, where X is a digit, work the way skypiax expects, while names like skyland-lineX don't work that way. I have no idea why. Nevertheless, even in this case, skypiax gets control over this call and can possibly manage it the way you would like. It just doesn't know what to do with this kind of skype protocol signaling. seven-7 wrote: > > ... I haven't play it into deep and not sure if one skype > instance can transfer to another account or another instance on busy. > > -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22398443.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gmaruzz at celliax.org Sun Mar 8 08:35:06 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 8 Mar 2009 16:35:06 +0100 Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <22339162.post@talk.nabble.com> References: <22339162.post@talk.nabble.com> Message-ID: <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> Ciao Dmitry, The warnings are unharmful, I've just fixed them as per svn 12524, so you will not see them anymore. But it will change nothing if there is a problem (I mean, the warnings are not the problem and are not indicating a problem). I cannot reproduce the problem, but maybe is because of the "strange name problem". It would be of great help if you do, from the FS CLI: console loglevel 9 then reproduce the problem, and then attach (attach, not copy) *all* the debug output (since beginning) to the Jira issue: http://jira.freeswitch.org/browse/MODSKYPIAX-28 Ciao for now, gm 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, Mar 8, 2009 at 11:17 AM, rdmitry wrote: > > Hi all, > > I've got a strange problem with skypiax. I successfully installed freeswitch > revision 12408 with skypiax and configured 2 skype channels with different > names. When I try to call both names one by one or simultaneously, > everything goes fine. But when I try to place a second call to the same > skype name which is busy with the first call, I get the following message: > > 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 > skypiax_signaling_read() rev 12409[(nil)|37 ? ? ][WARNINGA ?372 ?][skypiax1 > ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized > > and second call can't get thru. I can hear call progress tones only. After > about 5 seconds the message > > ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 > skypiax_signaling_read() rev 12409[(nil)|37 ? ? ][WARNINGA ?372 ?][skypiax1 > ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized > > occurs and I can hear only silence after that. > > Does anybody know what might cause such a problem? > > I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) > > Any help would be very much appreciated. > > Best regards, Dmitry > > > > > > -- > View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nik.middleton at noblesolutions.co.uk Sun Mar 8 09:03:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Mar 2009 16:03:45 -0000 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: <49B3C0C9.7040901@gmx.net> References: <49B3C0C9.7040901@gmx.net> Message-ID: That's exactly what I was looking for, many thanks Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P GMX Sent: 08 March 2009 12:58 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Getting a sip trace on the console I use the ngrep tool on the OS console and write the output to a file: ngrep -d any port 5060 -W byline >outfile.txt Best regards Peter Nik Middleton schrieb: > > Hi Guys, > > > > I'm trying to debug some SIP messaging issues. Is there a way of > doing the Asterisk equivalent of SIP Debug so I can see what's being sent? > > > > Regards, > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From kawarod at laposte.net Sun Mar 8 23:48:01 2009 From: kawarod at laposte.net (rod) Date: Mon, 09 Mar 2009 10:48:01 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B13277.9090505@3c.co.uk> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> Message-ID: <49B4BBA1.4000109@laposte.net> Hi David, already tried this :p the pbm is that this doesn' modify the RPID header, but it adds a new one so that I have 2 RPID header in the SIP INVITE :( rod David Knell wrote: > Hi Rod, > > You can set it directly: > application="set"> ;screen=yes;privacy=off]]> > > > --Dave > >> using these functions like this did nothing on the SIP INVITE packet :'( >> >> seven wrote: >> >>> try >>> bridge >>> ({effective_caller_id_name >>> ="your_name",effective_caller_id_number="0000"}sofia/b-leg) >>> >>> On Mar 5, 2009, at 9:00 PM, rod wrote: >>> >>> >>> >>>> the A leg invite looks like this: >>>> From: "Anonymous" >>>> >>>> it has been rewritten like this: >>>> From: "Anonymous" >>>> >>>> rod >>>> >>>> rod wrote: >>>> >>>> >>>>> Hi Brian, >>>>> >>>>> if I use the function effective_caller_id_number with my INVITE, I >>>>> get this: >>>>> >>>>> From: "Anonymous" ;tag=17geyFjX5p0gS. >>>>> >>>>> this is not exactly what I'm looking for :p >>>>> >>>>> rod >>>>> >>>>> >>>>> Brian West wrote: >>>>> >>>>> >>>>> >>>>>> Well this depends on how you're placing the call.. if its a standard >>>>>> bridge you can on the A-Leg set >>>>>> "effective_caller_id_number=000${caller_id_number}" before you call >>>>>> bridge. >>>>>> >>>>>> Is the from already in the correct format? >>>>>> >>>>>> /b >>>>>> >>>>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Dear list, >>>>>>> >>>>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>>>> only in >>>>>>> this header. >>>>>>> >>>>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>>>> this >>>>>>> example) in the RPID header : >>>>>>> >>>>>>> From: Anonymous;tag=1208367 >>>>>>> Remote-Party-ID: >>>>>>> >>>>>> 123456 >>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>> >>>>>>> should become: >>>>>>> From: Anonymous;tag=1208367 >>>>>>> Remote-Party-ID: >>>>>>> >>>>>> 000123456 >>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>> >>>>>>> But the From field should remain unchanged. >>>>>>> >>>>>>> And how to strip this prefix: >>>>>>> From: Anonymous;tag=1208367 >>>>>>> Remote-Party-ID: >>>>>>> >>>>>> 000123456 >>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>> >>>>>>> should become: >>>>>>> From: Anonymous;tag=1208367 >>>>>>> Remote-Party-ID: >>>>>>> >>>>>> 123456 >>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>> >>>>>>> >>>>>>> regards. >>>>>>> >>>>>>> >>>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rdmitry0911 at yandex.ru Mon Mar 9 01:32:18 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Mon, 9 Mar 2009 01:32:18 -0700 (PDT) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> References: <22339162.post@talk.nabble.com> <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> Message-ID: <22408941.post@talk.nabble.com> Hi Giovanni, I put everything you aked for in archive and attached it to the bug report at http://jira.freeswitch.org/browse/MODSKYPIAX-28 Hope it'll help to resolve this issue. Best Regards, Dmitry Giovanni Maruzzelli-3 wrote: > > Ciao Dmitry, > > The warnings are unharmful, I've just fixed them as per svn 12524, so > you will not see them anymore. But it will change nothing if there is > a problem (I mean, the warnings are not the problem and are not > indicating a problem). > > I cannot reproduce the problem, but maybe is because of the "strange > name problem". > > It would be of great help if you do, from the FS CLI: > > console loglevel 9 > > then reproduce the problem, and then attach (attach, not copy) *all* > the debug output (since beginning) to the Jira issue: > http://jira.freeswitch.org/browse/MODSKYPIAX-28 > > Ciao for now, > gm > > > > 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, Mar 8, 2009 at 11:17 AM, rdmitry wrote: >> >> Hi all, >> >> I've got a strange problem with skypiax. I successfully installed >> freeswitch >> revision 12408 with skypiax and configured 2 skype channels with >> different >> names. When I try to call both names one by one or simultaneously, >> everything goes fine. But when I try to place a second call to the same >> skype name which is busy with the first call, I get the following >> message: >> >> 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 >> skypiax_signaling_read() rev 12409[(nil)|37 ? ? ][WARNINGA ?372 >> ?][skypiax1 >> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized >> >> and second call can't get thru. I can hear call progress tones only. >> After >> about 5 seconds the message >> >> ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 >> skypiax_signaling_read() rev 12409[(nil)|37 ? ? ][WARNINGA ?372 >> ?][skypiax1 >> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized >> >> occurs and I can hear only silence after that. >> >> Does anybody know what might cause such a problem? >> >> I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) >> >> Any help would be very much appreciated. >> >> Best regards, Dmitry >> >> >> >> >> >> -- >> View this message in context: >> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22408941.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gmaruzz at celliax.org Mon Mar 9 01:37:04 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 9 Mar 2009 09:37:04 +0100 Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <22408941.post@talk.nabble.com> References: <22339162.post@talk.nabble.com> <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> <22408941.post@talk.nabble.com> Message-ID: <7b197bef0903090137k69e0acc4y3bffcfc3d364a65b@mail.gmail.com> Thank you Dmitry, I'll have a look into it this evening (6 hours from now :-) ) 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 Mon, Mar 9, 2009 at 9:32 AM, rdmitry wrote: > > Hi Giovanni, > > I put everything you aked for in archive and attached it to the bug report > at http://jira.freeswitch.org/browse/MODSKYPIAX-28 > > Hope it'll help to resolve this issue. > > Best Regards, Dmitry > > > Giovanni Maruzzelli-3 wrote: >> >> Ciao Dmitry, >> >> The warnings are unharmful, I've just fixed them as per svn 12524, so >> you will not see them anymore. But it will change nothing if there is >> a problem (I mean, the warnings are not the problem and are not >> indicating a problem). >> >> I cannot reproduce the problem, but maybe is because of the "strange >> name problem". >> >> It would be of great help if you do, from the FS CLI: >> >> console loglevel 9 >> >> then reproduce the problem, and then attach (attach, not copy) *all* >> the debug output (since beginning) to the Jira issue: >> http://jira.freeswitch.org/browse/MODSKYPIAX-28 >> >> Ciao for now, >> gm >> >> >> >> 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, Mar 8, 2009 at 11:17 AM, rdmitry wrote: >>> >>> Hi all, >>> >>> I've got a strange problem with skypiax. I successfully installed >>> freeswitch >>> revision 12408 with skypiax and configured 2 skype channels with >>> different >>> names. When I try to call both names one by one or simultaneously, >>> everything goes fine. But when I try to place a second call to the same >>> skype name which is busy with the first call, I get the following >>> message: >>> >>> 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 >>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>> ][skypiax1 >>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized >>> >>> and second call can't get thru. I can hear call progress tones only. >>> After >>> about 5 seconds the message >>> >>> ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 >>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>> ][skypiax1 >>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized >>> >>> occurs and I can hear only silence after that. >>> >>> Does anybody know what might cause such a problem? >>> >>> I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) >>> >>> Any help would be very much appreciated. >>> >>> Best regards, Dmitry >>> >>> >>> >>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22408941.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From diego.viola at gmail.com Sun Mar 8 19:30:01 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 8 Mar 2009 22:30:01 -0400 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: <49B3C0C9.7040901@gmx.net> Message-ID: <86a32abc0903081930g47c83f91h64e074bbe015c49b@mail.gmail.com> Use SVN, or wait for the next release, fs_cli+siptrace rocks :) On Sun, Mar 8, 2009 at 12:03 PM, Nik Middleton wrote: > That's exactly what I was looking for, many thanks > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Peter P GMX > Sent: 08 March 2009 12:58 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Getting a sip trace on the console > > I use the ngrep tool on the OS console and write the output to a file: > ngrep -d any port 5060 -W byline >outfile.txt > > Best regards > Peter > > Nik Middleton schrieb: >> >> Hi Guys, >> >> >> >> I'm trying to debug some SIP messaging issues. ?Is there a way of >> doing the Asterisk equivalent of SIP Debug so I can see what's being > sent? >> >> >> >> Regards, >> >> > ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From zhaoxxqq at 163.com Sun Mar 8 23:44:02 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Mon, 9 Mar 2009 14:44:02 +0800 Subject: [Freeswitch-users] Codec problems about G7221 with Polycom IP550 Message-ID: <200903091444010311430@163.com> Hello, I'm a newbe of Freeswitch. I have tried to config Polycom's soundpoint IP550 to use wideband codecs. G722 has no problem with conference and dial 9999. but with G7221 and G7221c have problems. I have config vars.xml to add and config polycom IP 550's SIP.cfg like the attachment. Can anyone help me to confirm if my config is right. Zhao Xiaoqiang 2009-03-09 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/a936c825/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sip.cfg Type: application/octet-stream Size: 183561 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/a936c825/attachment-0001.obj From brian at freeswitch.org Mon Mar 9 06:37:10 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Mar 2009 08:37:10 -0500 Subject: [Freeswitch-users] Codec problems about G7221 with Polycom IP550 In-Reply-To: <200903091444010311430@163.com> References: <200903091444010311430@163.com> Message-ID: <3A52AFA3-1A00-4AC4-8D1B-E8F92745693A@freeswitch.org> The IP550, 650 and 670 DO NOT support any G722.1 codecs at this point... expect support for those later in the year... right now they only support G722. /b On Mar 9, 2009, at 1:44 AM, zhaoxxqq wrote: > Hello, > I'm a newbe of Freeswitch. I have tried to config Polycom's > soundpoint IP550 to use wideband codecs. G722 has no problem with > conference and dial 9999. but with G7221 and G7221c have problems. > I have config vars.xml to add data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h"/> and config > polycom IP 550's SIP.cfg like the attachment. > Can anyone help me to confirm if my config is right. > > Zhao Xiaoqiang > > 2009-03-09 > zhaoxxqq > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/dbfae3b4/attachment.html From anthony.minessale at gmail.com Mon Mar 9 08:15:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Mar 2009 10:15:06 -0500 Subject: [Freeswitch-users] Freeswitch IAX support In-Reply-To: References: Message-ID: <191c3a030903090815l48670058u7a388c31adfe7304@mail.gmail.com> Hi, This issue is that our mod_iax is using the only freely available iax2 stack. A client library that was only designed for small softphones. There is a patch in jira to add registration support but it was not done correctly and we have not had much time to work on it. We've already had to add several unappealing hacks to the code we are using now to make it threadsafe and i don't think it will scale very far and you may find it a disappointment even with registration support. I had suggested at some point that we would consider making an entire new scalable implementation of iax2 designed as a client/server library but really it's probably the place of the authors of the protocol to provide such a resource. But if they do no wish to, I estimated the cost of developing such a stack to be in the range of 25k-35k. So the short answer is we have little to no demand for it, so we have not put much effort into supporting it. On Sun, Mar 8, 2009 at 7:11 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > Now that IAX has been published as an RFC ( > http://www.rfc-editor.org/authors/rfc5456.txt) are there any plans to > support registrations? > > > > Not a moan, just curious as to the road map. > > > > A lot of my users have Asterisk PBX?s using IAX and I?d love to replace my > Asterisk central server with FS to better serve them. Yes I know I could get > them to move to using SIP, but there?s a lot of them. > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/b466b87c/attachment.html From anthony.minessale at gmail.com Mon Mar 9 08:18:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Mar 2009 10:18:01 -0500 Subject: [Freeswitch-users] mod_nibblebill question: DB Error while updating cash! In-Reply-To: <86a32abc0903072228x67939d14x81a0c5e692cf67ac@mail.gmail.com> References: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> <86a32abc0903072228x67939d14x81a0c5e692cf67ac@mail.gmail.com> Message-ID: <191c3a030903090818n1694d93by7433280dc65424d0@mail.gmail.com> that means you should report it to jira not the mailing list. On Sun, Mar 8, 2009 at 1:28 AM, Diego Viola wrote: > Oh, I noticed the billing actually works, it discounts from my credit > but I still get that message, even if the update works. > > "2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error > while updating cash!" > > Thanks. > > On Sun, Mar 8, 2009 at 3:41 AM, Diego Viola wrote: > > Hello, > > > > I'm trying to give mod_nibblebill a try, I compiled it and created the > > DB, set up ODBC, etc. I'm using MySQL. > > > > This is how I created the db: > > > > CREATE TABLE accounts > > ( > > id int NOT NULL PRIMARY KEY, > > name VARCHAR(255), > > cash double precision NOT NULL > > ); > > > > However when I try to make a call I get this: > > > > 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error > > while updating cash! > > > > I have this also on my user directory: > > > > > > > > > > > > Any ideas? > > > > Thanks. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/e989a275/attachment.html From helmut.kuper at ewetel.de Mon Mar 9 08:19:19 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Mar 2009 16:19:19 +0100 Subject: [Freeswitch-users] Missing Diversion header in INVITE after 302 reply Message-ID: <49B53377.4040005@ewetel.de> Hello, following scenario: -Phone A is redirected unconditionally to phone C -Phone B calls A -Phone A replys with 302 and Dieversion header -FS detects the 302 and sends out a new INVITE to C I found that FS doesnt' include the received diversion sip header into the new INVITE sent to phone C. Is there a way to configure FS so that diversion header are included? Additionally: Is there a way to inform phone A about the diversion header, so that phone A get display a hint to user? regards Helmut From anthony.minessale at gmail.com Mon Mar 9 09:51:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Mar 2009 11:51:22 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <114ACB3A-3637-4FEA-A539-165A34DEAD09@ken-ton.com> Message-ID: <191c3a030903090951n675089fbjcc25e4fdf6cb26d4@mail.gmail.com> Thank you, I appriciate that you get some benifits from our efforts. We only recommend irc because it's an easy-to-access multi user chat where we can put all of the people who need help in the same room in real time so they can help each other and we can help them. I tried not to get annoyed about the 2 cracks in previous posts from others about how old and outdated irc was. Everyone is entitled to their own cup of tea after all. But we don't have to retire protocols just because they are old? We still use SMTP and HTTP and FTP and don't mock them for their age. Conversely, I feel kinda the same way about the "web 2.0" farce where the same tired browser and js nightmares I faced in 1997 are now swept under the rug with a singe addition of a background http-get instead of actually re-inventing the wheel if you are going to bother calling it wheel 2.0 Yes you can do some cool new stuff, but not nearly as much as what you could have done in 10 years of effort towards a better way, too late now ;) But that's only my opinion i don't try to enstill it to anyone. On Sat, Mar 7, 2009 at 11:40 PM, Luis F Urrea wrote: > Same thing happened to me in regards IRC, I had not used it for years > before getting into FS, but as a total newbie I can say that IRC is really > good to get things going quickly and all those No rocket science questions > we have. I have had people helping realtime looking at pastebin's and > stuff, which is a really good thing. > > When one needs to elaborate on a question maybe mailing list/forum fits > better, but what I know in my experience is that eventually you will get the > answer thanks to the efforts of the community with the current set of tools. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/2f02133e/attachment-0001.html From sergey.kirillov at gmail.com Mon Mar 9 11:41:02 2009 From: sergey.kirillov at gmail.com (Sergey Kirillov) Date: Mon, 09 Mar 2009 20:41:02 +0200 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card Message-ID: <49B562BE.1080902@gmail.com> Hi everybody, I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work. Can somebody help to to configure it? Problem log (Incoming call): ------------ 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel OpenZAP/1:1/2360012 [7473c92a-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->2360012 in context default 2009-03-06 14:58:14 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer OpenZAP/1:1/2360012 to XML[1000 at default] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->1000 in context default 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/opt/freeswitch/recordings/80503820933.2009-03-06-14-58-14.wav 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:1000 at 192.168.122.1:5061;transport=udp [748a2ba2-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:1/2360012! 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:15 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:26 [NOTICE] Span:0 Q.921() I frame in invalid state ignored ----------- Here are my config files --- openzap.conf -- [span wanpipe BRI_1] name => BRI_1 trunk_type => bri b-channel => 1:1-2 d-channel => 1:3 --- openzap.conf.xml --- --- wanpipe1.conf --- [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] wp1aft1 = wanpipe1, auto, API, Comment wp1aft2 = wanpipe1, auto, API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 5 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_HW_DTMF = NO [wp1aft1] HDLC_STREAMING = NO ACTIVE_CH = 1-15.17-31 IDLE_FLAG = 0x7E MTU = 240 MRU = 240 DATA_MUX = NO TDMV_HWEC = NO [wp1aft2] HDLC_STREAMING = YES ACTIVE_CH = 16 MTU = 1500 MRU = 1500 DATA_MUX = NO TDMV_HWEC = NO From msc at freeswitch.org Mon Mar 9 12:17:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Mar 2009 12:17:19 -0700 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card In-Reply-To: <49B562BE.1080902@gmail.com> References: <49B562BE.1080902@gmail.com> Message-ID: <87f2f3b90903091217n1b611c6cp88978d2aa1a79ba2@mail.gmail.com> > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > wp1aft1 = wanpipe1, auto, API, Comment > wp1aft2 = wanpipe1, auto, API, Comment > > [wanpipe1] > CARD_TYPE ? ? ? ? = AFT > S514CPU ? ? ? ? ? = A > CommPort ? ? ? ? ?= PRI > AUTO_PCISLOT ? ? ?= NO > PCISLOT ? ? ? ? ? = 4 > PCIBUS ? ? ? ? ? ?= 5 > FE_MEDIA ? ? ? ? ?= E1 > FE_LCODE ? ? ? ? ?= HDB3 > FE_FRAME ? ? ? ? ?= CRC4 > FE_LINE ? ? ? ? ? = 1 > TE_CLOCK ? ? ? ? ?= NORMAL > TE_REF_CLOCK ? ? ?= 0 > TE_HIGHIMPEDANCE ?= NO > TE_RX_SLEVEL ? ? ?= 120 > LBO ? ? ? ? ? ? ? = 120OH > TE_SIG_MODE ? ? ? = CCS > FE_TXTRISTATE ? ? = NO > MTU ? ? ? ? ? ? ? = 1500 > UDPPORT ? ? ? ? ? = 9000 > TTL ? ? ? ? ? ? ? = 255 > IGNORE_FRONT_END ?= NO > TDMV_HW_DTMF ? ? ?= NO > > [wp1aft1] > HDLC_STREAMING ?= NO > ACTIVE_CH ? ? ? = 1-15.17-31 > IDLE_FLAG ? ? ? = 0x7E > MTU ? ? ? ? ? ? = 240 > MRU ? ? ? ? ? ? = 240 > DATA_MUX ? ? ? ?= NO > TDMV_HWEC ? ? ? = NO > > [wp1aft2] > HDLC_STREAMING ?= YES > ACTIVE_CH ? ? ? = 16 > MTU ? ? ? ? ? ? = 1500 > MRU ? ? ? ? ? ? = 1500 > DATA_MUX ? ? ? ?= NO > TDMV_HWEC ? ? ? = NO > I'm no BRI expert but it looks to me like your wanpipe is set up for E1/EuroISDN. Where did you get this setup information? -MC From anthony.minessale at gmail.com Mon Mar 9 12:18:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Mar 2009 14:18:46 -0500 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card In-Reply-To: <49B562BE.1080902@gmail.com> References: <49B562BE.1080902@gmail.com> Message-ID: <191c3a030903091218k320bf077i1ad0461ea58f4ec9@mail.gmail.com> it's not released yet, please wait for the announcement that it has been completed sometime in the next week or 2. On Mon, Mar 9, 2009 at 1:41 PM, Sergey Kirillov wrote: > Hi everybody, > > I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work. > > Can somebody help to to configure it? > > Problem log (Incoming call): > ------------ > 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() > Unhandled event 2 > 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() > Unhandled event 2 > 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() > Unhandled event 2 > 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel OpenZAP/1:1/2360012 > [7473c92a-0a4e-11de-9dc3-c56d4d411902] > 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 80503820933->2360012 in context default > 2009-03-06 14:58:14 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() Transfer OpenZAP/1:1/2360012 to > XML[1000 at default] > 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 80503820933->1000 in context default > 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx > XML features > 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 > > record_session::/opt/freeswitch/recordings/80503820933.2009-03-06-14-58-14.wav > 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf > XML features > 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel > sofia/internal/sip:1000 at 192.168.122.1:5061;transport=udp > [748a2ba2-0a4e-11de-9dc3-c56d4d411902] > 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state > 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state > 2009-03-06 14:58:14 [NOTICE] switch_ivr_originate.c:1588 > switch_ivr_originate() Pre-Answer OpenZAP/1:1/2360012! > 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state > 2009-03-06 14:58:14 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received > unhandled message 125 (0x7d) > 2009-03-06 14:58:15 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received > unhandled message 125 (0x7d) > 2009-03-06 14:58:26 [NOTICE] Span:0 Q.921() I frame in invalid state > ignored > ----------- > > > Here are my config files > > --- openzap.conf -- > [span wanpipe BRI_1] > name => BRI_1 > trunk_type => bri > b-channel => 1:1-2 > d-channel => 1:3 > > > --- openzap.conf.xml --- > > > > > > > > > > > > > > > > > > > > > > --- wanpipe1.conf --- > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > wp1aft1 = wanpipe1, auto, API, Comment > wp1aft2 = wanpipe1, auto, API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 4 > PCIBUS = 5 > FE_MEDIA = E1 > FE_LCODE = HDB3 > FE_FRAME = CRC4 > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > TE_HIGHIMPEDANCE = NO > TE_RX_SLEVEL = 120 > LBO = 120OH > TE_SIG_MODE = CCS > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_HW_DTMF = NO > > [wp1aft1] > HDLC_STREAMING = NO > ACTIVE_CH = 1-15.17-31 > IDLE_FLAG = 0x7E > MTU = 240 > MRU = 240 > DATA_MUX = NO > TDMV_HWEC = NO > > [wp1aft2] > HDLC_STREAMING = YES > ACTIVE_CH = 16 > MTU = 1500 > MRU = 1500 > DATA_MUX = NO > TDMV_HWEC = NO > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/192035e8/attachment.html From rdmitry0911 at yandex.ru Mon Mar 9 12:37:06 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Mon, 9 Mar 2009 12:37:06 -0700 (PDT) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <7b197bef0903090137k69e0acc4y3bffcfc3d364a65b@mail.gmail.com> References: <22339162.post@talk.nabble.com> <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> <22408941.post@talk.nabble.com> <7b197bef0903090137k69e0acc4y3bffcfc3d364a65b@mail.gmail.com> Message-ID: <22417574.post@talk.nabble.com> Hi Giovanni, Finally I was able to manage the problem. It was my fault. I didn't realized, that the value of the "name" parameter in this line: should strictly correspond to skype name you regester in startskype.sh script. It can not be arbitrary chosen, as I thought before. After I fixed that, everything works fine. I think you can put this point into skypiax page of freeswitch wiki. Best Regards, Dmitry Giovanni Maruzzelli-3 wrote: > > Thank you Dmitry, > > I'll have a look into it this evening (6 hours from now :-) ) > > > 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 Mon, Mar 9, 2009 at 9:32 AM, rdmitry wrote: >> >> Hi Giovanni, >> >> I put everything you aked for in archive and attached it to the bug >> report >> at http://jira.freeswitch.org/browse/MODSKYPIAX-28 >> >> Hope it'll help to resolve this issue. >> >> Best Regards, Dmitry >> >> >> Giovanni Maruzzelli-3 wrote: >>> >>> Ciao Dmitry, >>> >>> The warnings are unharmful, I've just fixed them as per svn 12524, so >>> you will not see them anymore. But it will change nothing if there is >>> a problem (I mean, the warnings are not the problem and are not >>> indicating a problem). >>> >>> I cannot reproduce the problem, but maybe is because of the "strange >>> name problem". >>> >>> It would be of great help if you do, from the FS CLI: >>> >>> console loglevel 9 >>> >>> then reproduce the problem, and then attach (attach, not copy) *all* >>> the debug output (since beginning) to the Jira issue: >>> http://jira.freeswitch.org/browse/MODSKYPIAX-28 >>> >>> Ciao for now, >>> gm >>> >>> >>> >>> 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, Mar 8, 2009 at 11:17 AM, rdmitry wrote: >>>> >>>> Hi all, >>>> >>>> I've got a strange problem with skypiax. I successfully installed >>>> freeswitch >>>> revision 12408 with skypiax and configured 2 skype channels with >>>> different >>>> names. When I try to call both names one by one or simultaneously, >>>> everything goes fine. But when I try to place a second call to the same >>>> skype name which is busy with the first call, I get the following >>>> message: >>>> >>>> 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 >>>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>>> ][skypiax1 >>>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized >>>> >>>> and second call can't get thru. I can hear call progress tones only. >>>> After >>>> about 5 seconds the message >>>> >>>> ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 >>>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>>> ][skypiax1 >>>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized >>>> >>>> occurs and I can hear only silence after that. >>>> >>>> Does anybody know what might cause such a problem? >>>> >>>> I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) >>>> >>>> Any help would be very much appreciated. >>>> >>>> Best regards, Dmitry >>>> >>>> >>>> >>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22408941.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22417574.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From diego.viola at gmail.com Mon Mar 9 13:44:50 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 9 Mar 2009 16:44:50 -0400 Subject: [Freeswitch-users] mod_nibblebill question: DB Error while updating cash! In-Reply-To: <191c3a030903090818n1694d93by7433280dc65424d0@mail.gmail.com> References: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> <86a32abc0903072228x67939d14x81a0c5e692cf67ac@mail.gmail.com> <191c3a030903090818n1694d93by7433280dc65424d0@mail.gmail.com> Message-ID: <86a32abc0903091344k316e4f81w2ae157a6c46d52fc@mail.gmail.com> Ok, done. Thanks. On Mon, Mar 9, 2009 at 11:18 AM, Anthony Minessale wrote: > that means you should report it to jira not the mailing list. > > > On Sun, Mar 8, 2009 at 1:28 AM, Diego Viola wrote: >> >> Oh, I noticed the billing actually works, it discounts from my credit >> but I still get that message, even if the update works. >> >> "2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error >> while updating cash!" >> >> Thanks. >> >> On Sun, Mar 8, 2009 at 3:41 AM, Diego Viola wrote: >> > Hello, >> > >> > I'm trying to give mod_nibblebill a try, I compiled it and created the >> > DB, set up ODBC, etc. I'm using MySQL. >> > >> > This is how I created the db: >> > >> > ?CREATE TABLE accounts >> > ?( >> > ? id int NOT NULL PRIMARY KEY, >> > ? name VARCHAR(255), >> > ? cash double precision NOT NULL >> > ?); >> > >> > However when I try to make a call I get this: >> > >> > 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error >> > while updating cash! >> > >> > I have this also on my user directory: >> > >> > ? ? ? >> > ? ? ? >> > >> > >> > Any ideas? >> > >> > Thanks. >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From switchserver at gmail.com Mon Mar 9 15:17:41 2009 From: switchserver at gmail.com (Harry FSwitch) Date: Mon, 9 Mar 2009 18:17:41 -0400 Subject: [Freeswitch-users] RPC and web admin panel for conference? Message-ID: Hi all, I'm looking to implement an admin panel much like the one used at http://conference.freeswitch.org. Now I obviously cannot login and see the "admin" part of the panel but I'm pretty sure whats in there. I have xml_rpc running and can connect via http and issue commands. I've searched the forum here and went through the wiki, found nothing that looked like a panel. I was hoping to find a panel I can just configure and implement. Does anyone have a php (I guess, seeing as I have a php server) panel they can share with me? I'm sure I can get it working for my system. The thought of attempting one on my own at THIS point seems daunting at best. Any help would be greatly appreciated! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/e87c0c41/attachment.html From willbelair at yahoo.com Mon Mar 9 18:36:02 2009 From: willbelair at yahoo.com (Will Smith) Date: Mon, 9 Mar 2009 18:36:02 -0700 (PDT) Subject: [Freeswitch-users] STUN error Message-ID: <85688.97806.qm@web53606.mail.re2.yahoo.com> Hi, I have the FS worked? perfectly? under NAT. And when I moved it to a server with public IP, things getting wrong. This is the error message that I got: 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] ? -- ? I tried whatever I can think of like; set the or but still got the error. Could you please give me some guide how to fix this. ? Thanks ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/375de187/attachment.html From brian at freeswitch.org Mon Mar 9 18:52:10 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Mar 2009 20:52:10 -0500 Subject: [Freeswitch-users] STUN error In-Reply-To: <85688.97806.qm@web53606.mail.re2.yahoo.com> References: <85688.97806.qm@web53606.mail.re2.yahoo.com> Message-ID: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> Sounds like DNS failure maybe... might wanna remove the ext-sip-ip and ext-rtp-ip setting out of external.xml to take care of that. west philadelfia born and raised? /b On Mar 9, 2009, at 8:36 PM, Will Smith wrote: > Hi, > I have the FS worked perfectly under NAT. And when I moved it to a > server with public IP, things getting wrong. > This is the error message that I got: > 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 > sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: > 3478 [Remote Address Error!] > > -- From jason at jasonjgw.net Mon Mar 9 18:57:54 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 10 Mar 2009 12:57:54 +1100 Subject: [Freeswitch-users] STUN error In-Reply-To: <85688.97806.qm@web53606.mail.re2.yahoo.com> References: <85688.97806.qm@web53606.mail.re2.yahoo.com> Message-ID: <20090310015754.GA11792@jdc.jasonjgw.net> Will Smith wrote: > I tried whatever I can think of like; > set the > or > but still got the error. > Could you please give me some guide how to fix this. Change external_sip_ip and external_rtp_ip settings in vars.xml or in your external SIP profile. By default these are configured to use stun:stun.freeswitch.org. From willbelair at yahoo.com Mon Mar 9 20:09:44 2009 From: willbelair at yahoo.com (Will Smith) Date: Mon, 9 Mar 2009 20:09:44 -0700 (PDT) Subject: [Freeswitch-users] STUN error In-Reply-To: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> Message-ID: <694251.83720.qm@web53610.mail.re2.yahoo.com> Thank you Brian, it works like a champ. ? Yes,?west philadelfia born and raised? --- On Mon, 3/9/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] STUN error To: freeswitch-users at lists.freeswitch.org Date: Monday, March 9, 2009, 6:52 PM Sounds like DNS failure maybe... might wanna remove the ext-sip-ip and ext-rtp-ip setting out of external.xml to take care of that. west philadelfia born and raised? /b On Mar 9, 2009, at 8:36 PM, Will Smith wrote: > Hi, > I have the FS worked perfectly under NAT. And when I moved it to a > server with public IP, things getting wrong. > This is the error message that I got: > 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 > sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: > 3478 [Remote Address Error!] > > -- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/68bcc2d5/attachment.html From brian at freeswitch.org Mon Mar 9 20:39:41 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Mar 2009 22:39:41 -0500 Subject: [Freeswitch-users] STUN error In-Reply-To: <694251.83720.qm@web53610.mail.re2.yahoo.com> References: <694251.83720.qm@web53610.mail.re2.yahoo.com> Message-ID: <6BB7B147-A162-4F5C-8D66-FD7BF95DEE9B@freeswitch.org> Small joke :P Do you get that a lot? /b On Mar 9, 2009, at 10:09 PM, Will Smith wrote: > Thank you Brian, it works like a champ. > > Yes, west philadelfia born and raised? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/d95a67a3/attachment.html From dujinfang at gmail.com Mon Mar 9 21:05:35 2009 From: dujinfang at gmail.com (seven) Date: Tue, 10 Mar 2009 12:05:35 +0800 Subject: [Freeswitch-users] Script parsing a TPORT_DUMP sip log file to Mysql Message-ID: Hi all, I wrote a ruby script. it works for me. The script is in /scripts/ contrib/seven/sip/. All ideas and suggestions are welcome! Comment in script: Now and then we need to look at sip traces to see want happened on a failed call. There are lots of ways to monitor sip messages. However, not all of them are convinient as we want. Let's say a simple example: FreeSWITCH :> originate sofia/gateways/carrier1/5550000|sofia/gateways/ carrier2/5550000|sofia/carrier3... It's hard to tell what happend if the call fails. Because it's different sip sessions. The idea is to group them in one super session and see what happend. I do this by adding an arbitary sip header to do cross reference. And by parse the sip messages to a DB we can easily show it as html. I even can build a simple graph based on the DB data: http://skitch.com/seven1240/b8xj2/voip-master-idapted You can easily add a sip header to INVITE by(I use x_interaction): FreeSWITCH :> originate {sip_h_x_interaction=TEST0001}sofia/ gateways/..... And I can get all the messages from DB: SELECT * FROM `sip_messages` WHERE (call_id IN (SELECT distinct call_id FROM sip_messages WHERE x_interaction = 'TEST0001')) ORDER BY created_at; There are two aproches to get sip packets: 1) tcpdump/tshark 2) FreeSWITCH I use the second. Note, there is no way to actually get sip messages from FS currently, but sofia-sip has the ability to log all sip messages to a disk file by using TPORT_DUMP And you can use this script to parse them to a DB. I know it hurt performance, but we don't have tons of traffic and you know there are only 5-10 messages for each sip call. While we get about 1G bytes each day in the sip log, most of them are OPTIONS/NOTIFY etc. I filtered them before inserting to DB, but it will be better if sofia- sip can filter that :) The script will monitor the log file and parse and insert to DB in real time. It's written in the Ruby on Rails framework, however, I think it can run out of Rails with or without modification. But you still need ruby and rubygems if you want to use it. on Ubuntu/Debian # apt-get install ruby rubygems # gem install mysql file-tail yaml It's just an idea, you may like to write your own tools to parse the sip log file. Also the log file need to be rotated regularly. And I think it maybe possible to store the log file on a memory disk, whatever... :) Best -Seven. From diego.viola at gmail.com Mon Mar 9 21:09:55 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 10 Mar 2009 00:09:55 -0400 Subject: [Freeswitch-users] FS and Billing, AGI emulator Message-ID: <86a32abc0903092109m4f840129o6b104a14a38cdf6f@mail.gmail.com> Hi, I wanted to use A2Billing on FS, but I noticed it uses some AGI stuff for dialling and to check how much credit the user has, etc. I heard you could use A2B by just importing the FS CDR data into it, but that wont work, so I come to the conclusion that I have no way of doing billing on FS yet. There is ASTPP but that's not complete yet, and I heard vague comments of doing an AGI emulator on top of the event socket on FS, how hard that would be? Is there a possibility to do that? How much would it cost? Thanks, Diego From sergey.kirillov at gmail.com Tue Mar 10 00:48:01 2009 From: sergey.kirillov at gmail.com (Sergey Kirillov) Date: Tue, 10 Mar 2009 09:48:01 +0200 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card Message-ID: <49B61B31.8050307@gmail.com> > > I'm no BRI expert but it looks to me like your wanpipe is set up for > E1/EuroISDN. Where did you get this setup information? > -MC > It is autoconfigured by wancfg From sergey.kirillov at gmail.com Tue Mar 10 00:53:11 2009 From: sergey.kirillov at gmail.com (Sergey Kirillov) Date: Tue, 10 Mar 2009 09:53:11 +0200 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card Message-ID: <49B61C67.2070003@gmail.com> Is there a development version that I can check now? On Mon, Mar 9, 2009, Anthony Minessale wrote: > it's not released yet, > > please wait for the announcement that it has been completed sometime in the > next week or 2. From kawarod at laposte.net Tue Mar 10 04:00:57 2009 From: kawarod at laposte.net (rod) Date: Tue, 10 Mar 2009 15:00:57 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B4BBA1.4000109@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> Message-ID: <49B64869.6080007@laposte.net> Hi all, it seems there is no way to do this :( It could be great to be able to: - decide if RPID should be present or not in the B leg for an outbound call - make RPID header fully customizable with variables - filter RPID for inbound call I saw that kokoska rokoska created a jira bounty for 50$: Make RPID SIP header optional I'll add 150$ for this if I could manage RPID as described above. Sorry to use mailing list for this, I'm unable to add a note on jira for this bounty. regards, rod rod wrote: > Hi David, > > already tried this :p > the pbm is that this doesn' modify the RPID header, but it adds a new > one so that I have 2 RPID header in the SIP INVITE :( > > rod > > David Knell wrote: > >> Hi Rod, >> >> You can set it directly: >> > application="set">> ;screen=yes;privacy=off]]> >> >> >> --Dave >> >> >>> using these functions like this did nothing on the SIP INVITE packet :'( >>> >>> seven wrote: >>> >>> >>>> try >>>> bridge >>>> ({effective_caller_id_name >>>> ="your_name",effective_caller_id_number="0000"}sofia/b-leg) >>>> >>>> On Mar 5, 2009, at 9:00 PM, rod wrote: >>>> >>>> >>>> >>>> >>>>> the A leg invite looks like this: >>>>> From: "Anonymous" >>>>> >>>>> it has been rewritten like this: >>>>> From: "Anonymous" >>>>> >>>>> rod >>>>> >>>>> rod wrote: >>>>> >>>>> >>>>> >>>>>> Hi Brian, >>>>>> >>>>>> if I use the function effective_caller_id_number with my INVITE, I >>>>>> get this: >>>>>> >>>>>> From: "Anonymous" ;tag=17geyFjX5p0gS. >>>>>> >>>>>> this is not exactly what I'm looking for :p >>>>>> >>>>>> rod >>>>>> >>>>>> >>>>>> Brian West wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Well this depends on how you're placing the call.. if its a standard >>>>>>> bridge you can on the A-Leg set >>>>>>> "effective_caller_id_number=000${caller_id_number}" before you call >>>>>>> bridge. >>>>>>> >>>>>>> Is the from already in the correct format? >>>>>>> >>>>>>> /b >>>>>>> >>>>>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> Dear list, >>>>>>>> >>>>>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>>>>> only in >>>>>>>> this header. >>>>>>>> >>>>>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>>>>> this >>>>>>>> example) in the RPID header : >>>>>>>> >>>>>>>> From: Anonymous;tag=1208367 >>>>>>>> Remote-Party-ID: >>>>>>>> >>>>>>> 123456 >>>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>>> >>>>>>>> should become: >>>>>>>> From: Anonymous;tag=1208367 >>>>>>>> Remote-Party-ID: >>>>>>>> >>>>>>> 000123456 >>>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>>> >>>>>>>> But the From field should remain unchanged. >>>>>>>> >>>>>>>> And how to strip this prefix: >>>>>>>> From: Anonymous;tag=1208367 >>>>>>>> Remote-Party-ID: >>>>>>>> >>>>>>> 000123456 >>>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>>> >>>>>>>> should become: >>>>>>>> From: Anonymous;tag=1208367 >>>>>>>> Remote-Party-ID: >>>>>>>> >>>>>>> 123456 >>>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>>> >>>>>>>> >>>>>>>> regards. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From kokoska.rokoska at post.cz Tue Mar 10 04:30:49 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 12:30:49 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B64869.6080007@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> Message-ID: <49B64F69.5070209@post.cz> rod napsal(a): > Hi all, > > it seems there is no way to do this :( > > It could be great to be able to: > - decide if RPID should be present or not in the B leg for an > outbound call > - make RPID header fully customizable with variables > - filter RPID for inbound call > > I saw that kokoska rokoska created a jira bounty for 50$: Make RPID SIP > header optional > > I'll add 150$ for this if I could manage RPID as described above. > Sorry to use mailing list for this, I'm unable to add a note on jira for > this bounty. > I really don't need to customize RPID (cause it is depricated header), but... Well let's add another 50 $ if it will work like you describe :-) Best regards, kokoska.rokoska From kawarod at laposte.net Tue Mar 10 05:13:18 2009 From: kawarod at laposte.net (rod) Date: Tue, 10 Mar 2009 16:13:18 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B64F69.5070209@post.cz> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> Message-ID: <49B6595E.6040700@laposte.net> Yes I know, it's deprecated but many peers still rely on this and P-Asserted-ID is not widely spread (my own experience). moreover if we could strip the RPID, we could write a new one, but It could be very convenient to get access to the fields in this header for manipulation. thanks for supporting this request :p kokoska rokoska wrote: > > > rod napsal(a): > >> Hi all, >> >> it seems there is no way to do this :( >> >> It could be great to be able to: >> - decide if RPID should be present or not in the B leg for an >> outbound call >> - make RPID header fully customizable with variables >> - filter RPID for inbound call >> >> I saw that kokoska rokoska created a jira bounty for 50$: Make RPID SIP >> header optional >> >> I'll add 150$ for this if I could manage RPID as described above. >> Sorry to use mailing list for this, I'm unable to add a note on jira for >> this bounty. >> >> > > I really don't need to customize RPID (cause it is depricated header), > but... Well let's add another 50 $ if it will work like you describe :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From Richard.Lamkin at mettoni.com Tue Mar 10 03:26:22 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Tue, 10 Mar 2009 10:26:22 -0000 Subject: [Freeswitch-users] FS stopped working when NIC connection bounced. Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804930748@nickel.mettonigroup.com> Hi All, FS stopped working when NIC connection bounced. Q1- Has anyone an explanation of what happened ? Q2 - Is there a way to configure FS not to flip to another IP on a lost network connection. Q3 - Should an IP changed event be logged at a higher level e.g. [CRITICAL] ? NIC2 [172.22.240.156] lost connection to LAN then Freeswitch flipped to NIC1[192.168.1.1]. Then NIC2 restored connection and Freeswitch flipped back to NIC2. When it flipped back no SIP connections were restored. All PBX phones lost registration and my gateways to remote switches did not recover. FS is running but not responding to CLI or SIP. There is nothing logged in the Windows event viewer at the time of the incident. Sorry but I was not running Wireshark when this happened. ========== Log extract ======== 2009-03-09 21:00:21 [INFO] mod_sofia.c:2785 general_event_handler() IP change detected [172.22.240.156]->[192.168.1.1] []->[] 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write lock external 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting for worker thread 2009-03-09 21:00:21 [NOTICE] sofia_glue.c:2923 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-03-09 21:00:21 [INFO] switch_time.c:656 switch_load_timezones() Timezone reloaded 530 definitions 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write lock internal 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting for worker thread 2009-03-09 21:11:21 [INFO] mod_sofia.c:2785 general_event_handler() IP change detected [192.168.1.1]->[172.22.240.156] []->[] 2009-03-09 21:11:21 [NOTICE] sofia_glue.c:2923 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-03-09 21:11:21 [INFO] switch_time.c:656 switch_load_timezones() Timezone reloaded 530 definitions There were no more logged events after 2009-03-09 21:11:21. The log file was checked at 2009-03-10 08:50 . All phones had lost registration permanently I tried the CLI with "sofia status" this failed to return anything and the CLI was no longer responsive. I did a CTR-C just to see what was alive [ VS2008 trapped the break and I selected continue] 2009-03-10 08:58:25 [WARNING] switch_scheduler.c:114 task_thread_loop() Task was executed late by 4 seconds 1 heartbeat (core) Still no CLI. I the killed the FS, restarted and it worked as normal ==== Background; I'm testing Freeswitch on Windows XP with Release 1.03 from the source release tar ball, running in VS2008 debug mode. The PC has two NIC's 1 - 192.168.1.1 static 2 - 172.22.240.156 DHCP I'm using Nortel switches CS1K and CS2K for my upstream gateways and Linksys SPA942 for PBX clients. I'm only using SIP <-> SIP and all sip connections are via 172.22.240.156, There are no SIP devices on 192.168.1.1. It is unlikely the event was caused by a DHCP lease renew. DHCP lease details for 172.22.240.156. Lease Obtained. . . . . . . . . . : 09 March 2009 15:27:28 Lease Expires . . . . . . . . . . : 17 March 2009 15:27:28 === Regards Richard Lamkin PS: I am a new to FS and I am very enthusiastic about it potential. ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/92ad4c3f/attachment-0001.html From kokoska.rokoska at post.cz Tue Mar 10 05:32:51 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 13:32:51 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B6595E.6040700@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> Message-ID: <49B65DF3.9060001@post.cz> rod napsal(a): > ... if we could strip the RPID, we could write a new one, but It > could be very convenient to get access to the fields in this header for > manipulation. > Yes, rod, this is exactly why I update bounty to $100 :-) Thank you very much, rod, for support! Best regards, kokoska.rokoska From anthony.minessale at gmail.com Tue Mar 10 06:16:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 08:16:19 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B65DF3.9060001@post.cz> References: <49AFC1C3.9030603@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> Message-ID: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Latest SVN: Send no extra caller id info: {sip_cid_type=none}sofia/default/user at example.com Send RPID (default) {sip_cid_type=rpid}sofia/default/user at example.com Send P-XXX-Identity {sip_cid_type=pid}sofia/default/user at example.com Send RPID with chosen content {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ user at example.com [ Show ? ] Anthony Minessale IIadded a comment - 10/Mar/09 07:59 AM Send no extra caller id info: {sip_cid_type=none}sofia/default/ user at example.com Send RPID (default) {sip_cid_type=rpid}sofia/default/ user at example.com Send P-XXX-Identity {sip_cid_type=pid}sofia/default/ user at example.com Send RPID with chosen content {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ user at example.com Send RPID with chosen content and privacy flags (+ delimited, none to clear all flags) {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/ user at example.com [ Show ? ] Anthony Minessale IIadded a comment - 10/Mar/09 07:59 AM Send no extra caller id info: {sip_cid_type=none}sofia/default/ user at example.com Send RPID (default) {sip_cid_type=rpid}sofia/default/ user at example.com Send P-XXX-Identity {sip_cid_type=pid}sofia/default/ user at example.com Send RPID with chosen content {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ user at example.com Also the privacy app on the inbound leg controls the remaining contents of the RPID and Privacy headers. On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska wrote: > > rod napsal(a): > > ... if we could strip the RPID, we could write a new one, but It > > could be very convenient to get access to the fields in this header for > > manipulation. > > > > Yes, rod, this is exactly why I update bounty to $100 :-) > > Thank you very much, rod, for support! > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/2b6b7657/attachment.html From anthony.minessale at gmail.com Tue Mar 10 06:22:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 08:22:50 -0500 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card In-Reply-To: <49B61B31.8050307@gmail.com> References: <49B61B31.8050307@gmail.com> Message-ID: <191c3a030903100622g358ea918rb76d328f329bfbdf@mail.gmail.com> As I have already stated, it will be added to SVN as soon as it's complete. On Tue, Mar 10, 2009 at 2:48 AM, Sergey Kirillov wrote: > > > > I'm no BRI expert but it looks to me like your wanpipe is set up for > > E1/EuroISDN. Where did you get this setup information? > > -MC > > > It is autoconfigured by wancfg > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/90ebe410/attachment.html From sicfslist at gmail.com Tue Mar 10 06:28:25 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 10 Mar 2009 08:28:25 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Message-ID: <35b355e90903100628m277265d7g1a9dfe1492dd2a6b@mail.gmail.com> Anthony, That is awesome. This is something that will be a BIG help. SDR > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/3008dc65/attachment.html From kokoska.rokoska at post.cz Tue Mar 10 06:36:44 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 14:36:44 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Message-ID: <49B66CEC.7060002@post.cz> Anthony Minessale napsal(a): > Latest SVN: > > Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > [ Show ? ] > Anthony Minessale II > added a > comment - 10/Mar/09 07:59 AM Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > Send RPID with chosen content and privacy flags (+ delimited, none to > clear all flags) > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/user at example.com > > > > [ Show ? ] > Anthony Minessale II > added a > comment - 10/Mar/09 07:59 AM Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > Also the privacy app on the inbound leg controls the remaining contents > of the RPID and Privacy headers. > Incredible speed - like usually :-) Thank you very much, Anthony, for your work! Where should I send my bucks? Best regards, kokoska.rokoska From anthony.minessale at gmail.com Tue Mar 10 07:19:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 09:19:42 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B66CEC.7060002@post.cz> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B66CEC.7060002@post.cz> Message-ID: <191c3a030903100719w4ec5b18ei6d39871758c56231@mail.gmail.com> The paypal button on the hompege will do ;) On Tue, Mar 10, 2009 at 8:36 AM, kokoska rokoska wrote: > > > > Anthony Minessale napsal(a): > > Latest SVN: > > > > Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com user at example.com> > > > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com user at example.com> > > > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com user at example.com> > > > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > [ Show ? ] > > Anthony Minessale II > > added a > > comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > > > Send RPID with chosen content and privacy flags (+ delimited, none to > > clear all flags) > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/ > user at example.com > > > > > > > > [ Show ? ] > > Anthony Minessale II > > added a > > comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > > > > > Also the privacy app on the inbound leg controls the remaining contents > > of the RPID and Privacy headers. > > > > Incredible speed - like usually :-) > > Thank you very much, Anthony, for your work! Where should I send my bucks? > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/6b471531/attachment-0001.html From anthony.minessale at gmail.com Tue Mar 10 07:23:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 09:23:21 -0500 Subject: [Freeswitch-users] FS stopped working when NIC connection bounced. In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804930748@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804930748@nickel.mettonigroup.com> Message-ID: <191c3a030903100723u7d87cd20t8710c41ece88d2d7@mail.gmail.com> conf/autoload_configs/sofia.conf.xml uncomment out the auto-restart. On Tue, Mar 10, 2009 at 5:26 AM, Richard Lamkin wrote: > Hi All, > > > > FS stopped working when NIC connection bounced. > > > > Q1- Has anyone an explanation of what happened ? > > Q2 - Is there a way to configure FS not to flip to another IP on a lost > network connection. > > Q3 ? Should an IP changed event be logged at a higher level e.g. > [CRITICAL] ? > > > > NIC2 [172.22.240.156] lost connection to LAN then Freeswitch flipped to > NIC1[192.168.1.1]. Then NIC2 restored connection and Freeswitch flipped back > to NIC2. When it flipped back no SIP connections were restored. All PBX > phones lost registration and my gateways to remote switches did not > recover. FS is running but not responding to CLI or SIP. There is nothing > logged in the Windows event viewer at the time of the incident. Sorry but I > was not running Wireshark when this happened. > > > > ========== Log extract ======== > > > > 2009-03-09 21:00:21 [INFO] mod_sofia.c:2785 general_event_handler() IP > change detected [172.22.240.156]->[192.168.1.1] []->[] > > 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write > lock external > > 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting > for worker thread > > 2009-03-09 21:00:21 [NOTICE] sofia_glue.c:2923 > sofia_glue_restart_all_profiles() Reload XML [Success] > > 2009-03-09 21:00:21 [INFO] switch_time.c:656 switch_load_timezones() > Timezone reloaded 530 definitions > > 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write > lock internal > > 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting > for worker thread > > 2009-03-09 21:11:21 [INFO] mod_sofia.c:2785 general_event_handler() IP > change detected [192.168.1.1]->[172.22.240.156] []->[] > > 2009-03-09 21:11:21 [NOTICE] sofia_glue.c:2923 > sofia_glue_restart_all_profiles() Reload XML [Success] > > 2009-03-09 21:11:21 [INFO] switch_time.c:656 switch_load_timezones() > Timezone reloaded 530 definitions > > > > There were no more logged events after 2009-03-09 21:11:21. The log file > was checked at 2009-03-10 08:50 . All phones had lost registration > permanently > > I tried the CLI with ?sofia status? this failed to return anything and the > CLI was no longer responsive. > > > > I did a CTR-C just to see what was alive [ VS2008 trapped the break and I > selected continue] > > > > 2009-03-10 08:58:25 [WARNING] switch_scheduler.c:114 task_thread_loop() > Task was executed late by 4 seconds 1 heartbeat (core) > > > > Still no CLI. > > > > I the killed the FS, restarted and it worked as normal > > > > ==== > > Background; I?m testing Freeswitch on Windows XP with Release 1.03 from the > source release tar ball, running in VS2008 debug mode. > > > > The PC has two NIC?s > > 1 - 192.168.1.1 static > > 2 - 172.22.240.156 DHCP > > > > I?m using Nortel switches CS1K and CS2K for my upstream gateways and > Linksys SPA942 for PBX clients. > > > > I?m only using SIP <-> SIP and all sip connections are via 172.22.240.156, > There are no SIP devices on 192.168.1.1. > > > > It is unlikely the event was caused by a DHCP lease renew. > > > > DHCP lease details for 172.22.240.156. > > Lease Obtained. . . . . . . . . . : 09 March 2009 15:27:28 > > Lease Expires . . . . . . . . . . : 17 March 2009 15:27:28 > > > > === > > > > Regards > > > > Richard Lamkin > > > > PS: I am a new to FS and I am very enthusiastic about it potential. > > > > > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/1e30cebf/attachment.html From kawarod at laposte.net Tue Mar 10 07:30:28 2009 From: kawarod at laposte.net (rod) Date: Tue, 10 Mar 2009 18:30:28 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Message-ID: <49B67984.30104@laposte.net> Hi Anthony, thanks for this but I'd like to know if it's possible also to change only the caller_id_name and caller_id_number without modifying the from header. ex: with the origination variables I get this From: "test" ;tag=X4v4Kvt1B2DQF Remote-Party-ID: "test" ;party=calling;screen=yes;privacy=off and in the case of an anonymous call where from is like this on A-leg: From: Anonymous ;tag=z07m5db13cb-cbs450547977-o-5273-368294925 it is changed like this on B-leg: From: "test" ;tag=X4v4Kvt1B2DQF Moreover, in case of a bridged call, could you add the possibility to pass as is from header for an anonymous call. Running current, I have this: on A-leg: From: "Anonymous";tag=c0a80101-fed0c on B-leg: From: "Anonymous" ;tag=tpjD57U2pyrFN and I'd like B-leg to match A-leg for anonymous call as stated in RFC, anonymous at anonymous.invalid is the proposed way to handle anonymous call, I'll add 50$ more for support of this last request. thanks for your reactivity. regards, rod Anthony Minessale wrote: > Latest SVN: > > Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > > > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > > > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > > > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > [ Show ? ] > Anthony Minessale II > added > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > Send RPID with chosen content and privacy flags (+ delimited, none to > clear all flags) > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/user at example.com > > > > [ Show ? ] > Anthony Minessale II > added > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > Also the privacy app on the inbound leg controls the remaining > contents of the RPID and Privacy headers. > > > On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska > > wrote: > > > rod napsal(a): > > ... if we could strip the RPID, we could write a new one, but It > > could be very convenient to get access to the fields in this > header for > > manipulation. > > > > Yes, rod, this is exactly why I update bounty to $100 :-) > > Thank you very much, rod, for support! > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kokoska.rokoska at post.cz Tue Mar 10 07:57:20 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 15:57:20 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100719w4ec5b18ei6d39871758c56231@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B66CEC.7060002@post.cz> <191c3a030903100719w4ec5b18ei6d39871758c56231@mail.gmail.com> Message-ID: <49B67FD0.5070105@post.cz> Done :-) kokoska.rokoska Anthony Minessale napsal(a): > The paypal button on the hompege will do ;) > From anthony.minessale at gmail.com Tue Mar 10 08:17:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 10:17:23 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B67984.30104@laposte.net> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> Message-ID: <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> ok if you are up to date you should be able to add {sip_from_uri=sip:anonymous at anonymous.invalid} to your dial string. On Tue, Mar 10, 2009 at 9:30 AM, rod wrote: > Hi Anthony, > > thanks for this but I'd like to know if it's possible also to change > only the caller_id_name and caller_id_number without modifying the from > header. > > ex: with the origination variables I get this > From: "test" > >;tag=X4v4Kvt1B2DQF > Remote-Party-ID: "test" > > >;party=calling;screen=yes;privacy=off > > and in the case of an anonymous call where from is like this on A-leg: > From: Anonymous > >;tag=z07m5db13cb-cbs450547977-o-5273-368294925 > > it is changed like this on B-leg: > From: "test" > >;tag=X4v4Kvt1B2DQF > > Moreover, in case of a bridged call, could you add the possibility to > pass as is from header for an anonymous call. > Running current, I have this: > on A-leg: > From: > "Anonymous";tag=c0a80101-fed0c > on B-leg: > From: "Anonymous" > >;tag=tpjD57U2pyrFN > > and I'd like B-leg to match A-leg for anonymous call as stated in RFC, > anonymous at anonymous.invalid is the proposed way to handle anonymous > call, I'll add 50$ more for support of this last request. > > thanks for your reactivity. > regards, > rod > > Anthony Minessale wrote: > > Latest SVN: > > > > Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > > > > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > > > > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > > > > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > [ Show ? ] > > Anthony Minessale II > > added > > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > > > Send RPID with chosen content and privacy flags (+ delimited, none to > > clear all flags) > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/ > user at example.com > > > > > > > > [ Show ? ] > > Anthony Minessale II > > added > > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > > > > > Also the privacy app on the inbound leg controls the remaining > > contents of the RPID and Privacy headers. > > > > > > On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska > > > wrote: > > > > > > rod napsal(a): > > > ... if we could strip the RPID, we could write a new one, but It > > > could be very convenient to get access to the fields in this > > header for > > > manipulation. > > > > > > > Yes, rod, this is exactly why I update bounty to $100 :-) > > > > Thank you very much, rod, for support! > > > > Best regards, > > > > kokoska.rokoska > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/8bff0048/attachment-0001.html From kokoska.rokoska at post.cz Tue Mar 10 08:53:56 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 16:53:56 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> Message-ID: <49B68D14.5090908@post.cz> Anthony Minessale napsal(a): > ok if you are up to date you should be able to add > > {sip_from_uri=sip:anonymous at anonymous.invalid} to your dial string. > > Many thanks, Anthony, for that feature! It makes my life a lot easier :-) Best regards, kokoska.rokoska From helmut.kuper at ewetel.de Tue Mar 10 09:00:15 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 10 Mar 2009 17:00:15 +0100 Subject: [Freeswitch-users] Missing Diversion header in INVITE after 302 reply In-Reply-To: <49B53377.4040005@ewetel.de> References: <49B53377.4040005@ewetel.de> Message-ID: <49B68E8F.1010803@ewetel.de> Hello, has anybody an idea? regards Helmut On 09.03.2009 16:19, Helmut Kuper wrote: > Hello, > > following scenario: > > -Phone A is redirected unconditionally to phone C > -Phone B calls A > -Phone A replys with 302 and Dieversion header > -FS detects the 302 and sends out a new INVITE to C > > I found that FS doesn't include the received diversion sip header into > the new INVITE sent to phone C. > > Is there a way to configure FS so that diversion header are included? > > Additionally: Is there a way to inform phone A about the diversion > header, so that phone A get display a hint to user? > > regards > Helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From anthony.minessale at gmail.com Tue Mar 10 10:18:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 12:18:07 -0500 Subject: [Freeswitch-users] IRC pre release party Message-ID: <191c3a030903101018s3e8e0c3ai27cec27f7bb8ba23@mail.gmail.com> If you have nothing better to do drop by IRC We are up to 193 users and about to cross 200 for the first time. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/1302c16d/attachment.html From msc at freeswitch.org Tue Mar 10 13:48:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Mar 2009 13:48:08 -0700 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Message-ID: <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale wrote: > Latest SVN: > > Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content >{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul tuser at example.com FYI, I added this info to the channel variables page: http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type -MC From Prometheus001 at gmx.net Tue Mar 10 15:20:39 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 10 Mar 2009 23:20:39 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> Message-ID: <49B6E7B7.1090003@gmx.net> Hello, are these variables only available at call setup time or can they be changed during a call, e.g. before a call is being transferred to another destination? Best regards Peter Michael Collins schrieb: > On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale > wrote: > >> Latest SVN: >> >> Send no extra caller id info: >> {sip_cid_type=none}sofia/default/user at example.com >> >> Send RPID (default) >> {sip_cid_type=rpid}sofia/default/user at example.com >> >> Send P-XXX-Identity >> {sip_cid_type=pid}sofia/default/user at example.com >> >> Send RPID with chosen content >> {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul tuser at example.com >> > > FYI, I added this info to the channel variables page: > http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From krice at suspicious.org Tue Mar 10 15:29:30 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 10 Mar 2009 17:29:30 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B6E7B7.1090003@gmx.net> Message-ID: These should be available any time you are going to process a call thru the dialplan and call a bridge on the call > From: Peter P GMX > Reply-To: > Date: Tue, 10 Mar 2009 23:20:39 +0100 > To: > Subject: Re: [Freeswitch-users] Rewriting Remote Party ID > > Hello, > > are these variables only available at call setup time or can they be > changed during a call, e.g. before a call is being transferred to > another destination? > > Best regards > Peter > > Michael Collins schrieb: >> On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale >> wrote: >> >>> Latest SVN: >>> >>> Send no extra caller id info: >>> {sip_cid_type=none}sofia/default/user at example.com >>> >>> Send RPID (default) >>> {sip_cid_type=rpid}sofia/default/user at example.com >>> >>> Send P-XXX-Identity >>> {sip_cid_type=pid}sofia/default/user at example.com >>> >>> Send RPID with chosen content >>> {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_num >>> ber=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul >>> tuser at example.com >>> >> >> FYI, I added this info to the channel variables page: >> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type >> >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at suspicious.org Tue Mar 10 15:32:53 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 10 Mar 2009 17:32:53 -0500 Subject: [Freeswitch-users] RPC and web admin panel for conference? In-Reply-To: Message-ID: Hey, I just implemented something like this and commited it to my contrib directory (scripts/contrib/swk ) its a mixture of amf-php, ESL, and Flex... Its not complete by anymeans and you need Flex3 to compile the UI... Anyone wanting to throw some patches at it for other functionality are welcome to do so... One thing is severly lacks at this time is ANY sort of authentication...so you wouldn?t want it publically open to the world K From: Harry FSwitch Reply-To: Date: Mon, 9 Mar 2009 18:17:41 -0400 To: Subject: [Freeswitch-users] RPC and web admin panel for conference? Hi all, I'm looking to implement an admin panel much like the one used at http://conference.freeswitch.org. Now I obviously cannot login and see the "admin" part of the panel but I'm pretty sure whats in there. I have xml_rpc running and can connect via http and issue commands. I've searched the forum here and went through the wiki, found nothing that looked like a panel. I was hoping to find a panel I can just configure and implement. Does anyone have a php (I guess, seeing as I have a php server) panel they can share with me? I'm sure I can get it working for my system. The thought of attempting one on my own at THIS point seems daunting at best. Any help would be greatly appreciated! Thanks _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/69a3a6f0/attachment.html From mszlazak at aol.com Tue Mar 10 23:46:40 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 11 Mar 2009 02:46:40 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. Message-ID: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> I have the following in my dialplan. Individually, each extension does what it's suppose to do when dialing 1111. However, if I place continue=true in the first extension then it alone gets executed and the succeeding extension does not. I thought condition=true would allow the extension afterward to execute. My test hardware for this dialplan is a single PSTN line. A call comes in that line and "myExtension" executes then hopefully hangs up to free the line. Afterward, I want "myExtension_Continued" to execute the .js application and dial out that single PSTN line. I need help in getting this scenerio to work. Thanks. ?? ? ?? ??? ? ?? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ? ?? ? ?? ??? ? ?? ??? ? ?? ??? ??? ? ?? ??? ? ?? ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/e6dc106f/attachment-0001.html From helmut.kuper at ewetel.de Wed Mar 11 01:07:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Mar 2009 09:07:02 +0100 Subject: [Freeswitch-users] Waiting for playing a file as early media Message-ID: <49B77126.2030500@ewetel.de> Hello, I'm looking for a way to play a file exactly once in early media state of a call and then do e.g. a 302 reply. When I try it via ring_ready and ringback The playback immediately stops, when I send a respond. regards Helmut From kawarod at laposte.net Wed Mar 11 01:33:33 2009 From: kawarod at laposte.net (rod) Date: Wed, 11 Mar 2009 12:33:33 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> Message-ID: <49B7775D.3030308@laposte.net> thanks a lot Anthony, I consider the work is done and I did the paypal transfer of 200$. But I'll be happy if you could do some minor changes: I used this in the dialplan: and I get this From: "test" ;tag=tQ52gmv6NyN0D. Remote-Party-ID: "test" ;party=calling;screen=yes;privacy=off. as you may see, the origination_caller_id_name has been put in the from URI, and the domain part in the remote party ID has been transformed to @anonymous.invalid. It would be perfect if I could get this instead: From: "Anonymous" ;tag=tQ52gmv6NyN0D. Remote-Party-ID: "test" ;party=calling;screen=yes;privacy=off. where 172.29.0.5 is the external IP used by FS to bridge the call in this example (but I'm sure you already know this :p) regards, rod Anthony Minessale wrote: > ok if you are up to date you should be able to add > > {sip_from_uri=sip:anonymous at anonymous.invalid} to your dial string. > > > On Tue, Mar 10, 2009 at 9:30 AM, rod > wrote: > > Hi Anthony, > > thanks for this but I'd like to know if it's possible also to change > only the caller_id_name and caller_id_number without modifying the > from > header. > > ex: with the origination variables I get this > From: "test" >;tag=X4v4Kvt1B2DQF > Remote-Party-ID: "test" > >;party=calling;screen=yes;privacy=off > > and in the case of an anonymous call where from is like this on A-leg: > From: Anonymous > ;tag=z07m5db13cb-cbs450547977-o-5273-368294925 > > it is changed like this on B-leg: > From: "test" >;tag=X4v4Kvt1B2DQF > > Moreover, in case of a bridged call, could you add the possibility to > pass as is from header for an anonymous call. > Running current, I have this: > on A-leg: > From: > "Anonymous";tag=c0a80101-fed0c > on B-leg: > From: "Anonymous" >;tag=tpjD57U2pyrFN > > and I'd like B-leg to match A-leg for anonymous call as stated in RFC, > anonymous at anonymous.invalid is the proposed way to handle anonymous > call, I'll add 50$ more for support of this last request. > > thanks for your reactivity. > regards, > rod > > Anthony Minessale wrote: > > Latest SVN: > > > > Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > > > > > > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > > > > > > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > > > > > > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > > > [ Show ? ] > > Anthony Minessale II > > > added > > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > > > Send RPID > (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > > > Send > P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > > > Send RPID > with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > > > > > Send RPID with chosen content and privacy flags (+ delimited, > none to > > clear all flags) > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/user at example.com > > > > > > > > > > [ Show ? ] > > Anthony Minessale II > > > added > > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > > > Send RPID > (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > > > Send > P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > > > Send RPID > with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > > > > > > > Also the privacy app on the inbound leg controls the remaining > > contents of the RPID and Privacy headers. > > > > > > On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska > > > >> > wrote: > > > > > > rod napsal(a): > > > ... if we could strip the RPID, we could write a new one, > but It > > > could be very convenient to get access to the fields in this > > header for > > > manipulation. > > > > > > > Yes, rod, this is exactly why I update bounty to $100 :-) > > > > Thank you very much, rod, for support! > > > > Best regards, > > > > kokoska.rokoska > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shanwlin at gmail.com Tue Mar 10 18:17:20 2009 From: shanwlin at gmail.com (shawn lin) Date: Wed, 11 Mar 2009 09:17:20 +0800 Subject: [Freeswitch-users] Where does the FS keep the call-id list? Message-ID: <9fcf45ed0903101817t3565832te8b166693e1ddbaa@mail.gmail.com> Hi all, I am new to FreeSwitch, I have a question when using SIPp <-> FreeSwitch <-> SIPp. The SIPp UAC create many calls and the SIPp UAS does reveive them. I wonder if anyone can tell me, where does the FreeSwitch keep the call-ids? Is there a list to contain all the call-id? If there is a list, how can I found it? Best Regards! Shawn Lin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/ee15d70f/attachment.html From mike at jerris.com Wed Mar 11 04:34:38 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Mar 2009 07:34:38 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> Message-ID: The continue works fine, it just hangs up befoer that due to: Mike On Mar 11, 2009, at 2:46 AM, mszlazak at aol.com wrote: > I have the following in my dialplan. > > Individually, each extension does what it's suppose to do when > dialing 1111. > However, if I place continue=true in the first extension then it > alone gets executed and the succeeding extension does not. > I thought condition=true would allow the extension afterward to > execute. > > My test hardware for this dialplan is a single PSTN line. > A call comes in that line and "myExtension" executes then hopefully > hangs up to free the line. > Afterward, I want "myExtension_Continued" to execute the .js > application and dial out that single PSTN line. > > I need help in getting this scenerio to work. > > Thanks. > > > > > > data="effective_caller_id_number=${caller_id_number}"/> > data="hangup_after_bridge=true"/> > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/c33d4357/attachment.html From mike at jerris.com Wed Mar 11 04:36:47 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Mar 2009 07:36:47 -0400 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <49B77126.2030500@ewetel.de> References: <49B77126.2030500@ewetel.de> Message-ID: <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> If your actually playing back a file (that blocks) it should work fine. Ringback goes in the background so it doesn't wait for it to "finish" before it does the ringback Mike On Mar 11, 2009, at 4:07 AM, Helmut Kuper wrote: > Hello, > > I'm looking for a way to play a file exactly once in early media state > of a call and then do e.g. a 302 reply. When I try it via ring_ready > and > ringback The playback immediately stops, when I send a respond. > > regards > Helmut > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Wed Mar 11 05:22:43 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Mar 2009 13:22:43 +0100 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> References: <49B77126.2030500@ewetel.de> <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> Message-ID: <49B7AD13.3040903@ewetel.de> Hi Mike, Thank you! regards Helmut On 11.03.2009 12:36, Michael Jerris wrote: > If your actually playing back a file (that blocks) it should work > fine. Ringback goes in the background so it doesn't wait for it to > "finish" before it does the ringback From anthony.minessale at gmail.com Wed Mar 11 06:16:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Mar 2009 08:16:38 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B7775D.3030308@laposte.net> References: <49AFC1C3.9030603@laposte.net> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> Message-ID: <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> On Wed, Mar 11, 2009 at 3:33 AM, rod wrote: > thanks a lot Anthony, > > I consider the work is done and I did the paypal transfer of 200$. > > But I'll be happy if you could do some minor changes: > > Done....r12563 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/0c7211ba/attachment-0001.html From helmut.kuper at ewetel.de Wed Mar 11 06:19:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Mar 2009 14:19:38 +0100 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <49B7AD13.3040903@ewetel.de> References: <49B77126.2030500@ewetel.de> <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> <49B7AD13.3040903@ewetel.de> Message-ID: <49B7BA6A.6030108@ewetel.de> Hi again, it works good. Can I assign a sleep to playback somehow to avoid loose of phrases at the beginning? regards Helmut On 11.03.2009 13:22, Helmut Kuper wrote: > Hi Mike, > > Thank you! > > regards > Helmut > > > On 11.03.2009 12:36, Michael Jerris wrote: > >> If your actually playing back a file (that blocks) it should work >> fine. Ringback goes in the background so it doesn't wait for it to >> "finish" before it does the ringback >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From shahal at jajah.com Wed Mar 11 07:08:28 2009 From: shahal at jajah.com (Shahal Hazan) Date: Wed, 11 Mar 2009 16:08:28 +0200 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app during bridge between softphone ext 1000 and PSTN. Message-ID: Hi, After I call the external number successfully, I'm able to receive DTMF from the softphone but the PSTN's DTMF doesn't work. I am able to change the mute status during a conference from the same PSTN. I had a look at: http://wiki.freeswitch.org/wiki/Dtmf_troubleshooting but it didn't help. I have: on the Local_Extension and on my SIP_PROVIDER extension, both at sofia/default Any ideas? Shahal Hazan Mobile Team Jajah My blog: http://jajahdevblog.com/shahal www.Jajah.com M: +972-54-227-9567 This mail was sent via Mail-SeCure System. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/0647daf3/attachment.html From kawarod at laposte.net Wed Mar 11 07:09:30 2009 From: kawarod at laposte.net (rod) Date: Wed, 11 Mar 2009 18:09:30 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> Message-ID: <49B7C61A.6070201@laposte.net> ok, almost perfect :p using this in dialplan: {sip_cid_type=rpid,origination_caller_id_number=000000${CALLER_NUMBER},origination_caller_id_name=test,originatio\ n_privacy=screen+hide_name+hide_number,sip_from_uri=sip:anonymous at anonymous.invalid}sofia/ss7/000000${destination_number}@${GW_IP} I have this on B leg From: "test" ;tag=gQ53Hjp617BSH. Remote-Party-ID: "test" ;party=calling;screen=yes;privacy=off. As you can see the caller id name is still modified, do you think you could add something like sip_from_uri_name to circumvent this. Anthony, I tried many settings for origination_privacy and it seems to do nothing on the RPID header. Any clue? regards. Anthony Minessale wrote: > > > On Wed, Mar 11, 2009 at 3:33 AM, rod > wrote: > > thanks a lot Anthony, > > I consider the work is done and I did the paypal transfer of 200$. > > But I'll be happy if you could do some minor changes: > > > Done....r12563 > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From miles.chet at gmail.com Wed Mar 11 07:15:57 2009 From: miles.chet at gmail.com (roberto) Date: Wed, 11 Mar 2009 11:15:57 -0300 Subject: [Freeswitch-users] suggestions to application Message-ID: Hello, I have an app write in php running with asterisk for calling cards, i would like to migrate everything to freeswitch, but i think that fs has almost everything that a calling card need read to use.. I?m looking for suggestion on some features like: Accounting Rate Billing Route Please someone could give me any suggestion on wich program that i should use. Thanks, -- "Without love, we are birds with broken wings." Morrie From anthony.minessale at gmail.com Wed Mar 11 08:14:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Mar 2009 10:14:36 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B7C61A.6070201@laposte.net> References: <49AFC1C3.9030603@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> Message-ID: <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> try now On Wed, Mar 11, 2009 at 9:09 AM, rod wrote: > ok, > > almost perfect :p > > using this in dialplan: > > {sip_cid_type=rpid,origination_caller_id_number=000000${CALLER_NUMBER},origination_caller_id_name=test,originatio\ > > n_privacy=screen+hide_name+hide_number,sip_from_uri=sip:anonymous at anonymous.invalid > }sofia/ss7/000000${destination_number}@${GW_IP} > > I have this on B leg > From: "test" ;tag=gQ53Hjp617BSH. > Remote-Party-ID: "test" > > >;party=calling;screen=yes;privacy=off. > > As you can see the caller id name is still modified, do you think you > could add something like sip_from_uri_name to circumvent this. > > Anthony, I tried many settings for origination_privacy and it seems to > do nothing on the RPID header. Any clue? > > regards. > > Anthony Minessale wrote: > > > > > > On Wed, Mar 11, 2009 at 3:33 AM, rod > > wrote: > > > > thanks a lot Anthony, > > > > I consider the work is done and I did the paypal transfer of 200$. > > > > But I'll be happy if you could do some minor changes: > > > > > > Done....r12563 > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/a24d4979/attachment.html From anthony.minessale at gmail.com Wed Mar 11 08:15:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Mar 2009 10:15:12 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> Message-ID: <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> oops hit send too soon, try now using sip_from_display variable to control the display field in the from url On Wed, Mar 11, 2009 at 10:14 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try now > > > On Wed, Mar 11, 2009 at 9:09 AM, rod wrote: > >> ok, >> >> almost perfect :p >> >> using this in dialplan: >> >> {sip_cid_type=rpid,origination_caller_id_number=000000${CALLER_NUMBER},origination_caller_id_name=test,originatio\ >> >> n_privacy=screen+hide_name+hide_number,sip_from_uri=sip:anonymous at anonymous.invalid >> }sofia/ss7/000000${destination_number}@${GW_IP} >> >> I have this on B leg >> From: "test" ;tag=gQ53Hjp617BSH. >> Remote-Party-ID: "test" >> >> >;party=calling;screen=yes;privacy=off. >> >> As you can see the caller id name is still modified, do you think you >> could add something like sip_from_uri_name to circumvent this. >> >> Anthony, I tried many settings for origination_privacy and it seems to >> do nothing on the RPID header. Any clue? >> >> regards. >> >> Anthony Minessale wrote: >> > >> > >> > On Wed, Mar 11, 2009 at 3:33 AM, rod > > > wrote: >> > >> > thanks a lot Anthony, >> > >> > I consider the work is done and I did the paypal transfer of 200$. >> > >> > But I'll be happy if you could do some minor changes: >> > >> > >> > Done....r12563 >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > >> > 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:213-799-1400 >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/40ee6a4c/attachment-0001.html From chris at maxpowersoft.com Wed Mar 11 08:25:57 2009 From: chris at maxpowersoft.com (Chris Danielson) Date: Wed, 11 Mar 2009 08:25:57 -0700 Subject: [Freeswitch-users] C API session and channels questions In-Reply-To: References: Message-ID: <49B7D805.50303@maxpowersoft.com> Hello Guys, I have a question regarding how I can do the following within the FS C API. What API can I use to list the current channels and sessions? What I have already in place is a generic FS module capable of serializing any sort of C data and storing it onto another device or caching server. I then have the ability to deserialize and operate on these generic objects already in my module. What I anticipate doing is finding a path to load up the FreeSWITCH (session and channel) objects and deserialize them into a freshly started FreeSWITCH instance. Any ideas on what switch_core_session.c methods, etc. I should use and any pitfalls? My primary focus for my company is conference calling, but I'd like to make this a means to work with everything that I possibly can. As soon as I get my basic testing and instances working I'll offer up the code to bkw, mikej and anthm for review. As-is, I'm not even sure if what I'm asking is even feasible, but I think having a module like this would be a good start to fill my needs and perhaps those of the community. So again, my two questions are as follows: 1) What API can I use to enumerate through the current channels and sessions initialized and running in memory? (I'm looking to clone and serialize them and store them on a separate server). 2) Any helper methods or ways that I can re-construct the channels and sessions into memory on a freshly started instance of FreeSWITCH? Kind Regards, Chris Danielson (aka. danchris) From msc at freeswitch.org Wed Mar 11 08:40:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 08:40:30 -0700 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <49B7BA6A.6030108@ewetel.de> References: <49B77126.2030500@ewetel.de> <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> <49B7AD13.3040903@ewetel.de> <49B7BA6A.6030108@ewetel.de> Message-ID: <87f2f3b90903110840p5ba9412dk8211377a4992fd3e@mail.gmail.com> > it works good. Can I assign a sleep to playback somehow to avoid loose > of phrases at the beginning? Are you playing just a single file? You can use the phrase macros to create a pause at the beginning of your playback. Or you can cheat and prepend a few hundred (or thousand) milliseconds of silence at the beginning of your sound file. :) -MC From msc at freeswitch.org Wed Mar 11 08:46:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 08:46:17 -0700 Subject: [Freeswitch-users] suggestions to application In-Reply-To: References: Message-ID: <87f2f3b90903110846j69d73a0ct58a4b41f3c06168e@mail.gmail.com> > I?m looking for suggestion on some features like: > > Accounting > Rate > Billing > Route There's nothing out-of-box ready, but you should look at these mods: mod_nibblebill - real-time billing mod_lcr - outbound routing mod_easyroute - inbound routing Start with the wiki: http:wiki.freeswitch.org There's a lot of research for you to do. Have fun! :) -MC From msc at freeswitch.org Wed Mar 11 08:51:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 08:51:06 -0700 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app during bridge between softphone ext 1000 and PSTN. In-Reply-To: References: Message-ID: <87f2f3b90903110851q1f1589bfq6102cf4c65c55130@mail.gmail.com> > After I call the external number successfully, I?m able to receive DTMF from > the softphone but the PSTN?s DTMF doesn?t work. We definitely don't want to assume anything, so I have to ask the obvious questions: who is the provider? are the DTMFs in-band or RFC2833? Any chance you can turn on full debugging and see if there are any clues? Thanks! -MC From msc at freeswitch.org Wed Mar 11 08:54:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 08:54:43 -0700 Subject: [Freeswitch-users] C API session and channels questions In-Reply-To: <49B7D805.50303@maxpowersoft.com> References: <49B7D805.50303@maxpowersoft.com> Message-ID: <87f2f3b90903110854x580eea6dkd42efab02c4c3ff8@mail.gmail.com> On Wed, Mar 11, 2009 at 8:25 AM, Chris Danielson wrote: > Hello Guys, > I have a question regarding how I can do the following within the FS C API. > The devs love it when people get down and dirty with FS! However, this thread is definitely better suited for the freeswitch-dev mailing list. Also, I notice that you're in IRC #freeswitch but not #freeswitch-dev. Hop in to the dev channel and you'll have a handful of people who will be willing to kick this stuff around with you. -MC From chris at maxpowersoft.com Wed Mar 11 09:07:26 2009 From: chris at maxpowersoft.com (Chris Danielson) Date: Wed, 11 Mar 2009 09:07:26 -0700 Subject: [Freeswitch-users] C API session and channels questions In-Reply-To: <87f2f3b90903110854x580eea6dkd42efab02c4c3ff8@mail.gmail.com> References: <49B7D805.50303@maxpowersoft.com> <87f2f3b90903110854x580eea6dkd42efab02c4c3ff8@mail.gmail.com> Message-ID: <49B7E1BE.8010504@maxpowersoft.com> Michael, Thanks! I'm joining up now. Regards, Chris Michael Collins wrote: > On Wed, Mar 11, 2009 at 8:25 AM, Chris Danielson wrote: > >> Hello Guys, >> I have a question regarding how I can do the following within the FS C API. >> >> > > The devs love it when people get down and dirty with FS! However, this > thread is definitely better suited for the freeswitch-dev mailing > list. Also, I notice that you're in IRC #freeswitch but not > #freeswitch-dev. Hop in to the dev channel and you'll have a handful > of people who will be willing to kick this stuff around with you. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/77a0ad64/attachment.html From helmut.kuper at ewetel.de Wed Mar 11 09:31:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Mar 2009 17:31:38 +0100 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <87f2f3b90903110840p5ba9412dk8211377a4992fd3e@mail.gmail.com> References: <49B77126.2030500@ewetel.de> <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> <49B7AD13.3040903@ewetel.de> <49B7BA6A.6030108@ewetel.de> <87f2f3b90903110840p5ba9412dk8211377a4992fd3e@mail.gmail.com> Message-ID: <49B7E76A.5030901@ewetel.de> Hi Mike, yes, it's just one single file. I had the same idea to cheat a bit, but a dynamic solution is better ... will test the phrase macros. thx for the hint! regards helmut On 11.03.2009 16:40, Michael Collins wrote: >> it works good. Can I assign a sleep to playback somehow to avoid loose >> of phrases at the beginning? >> > Are you playing just a single file? You can use the phrase macros to > create a pause at the beginning of your playback. Or you can cheat and > prepend a few hundred (or thousand) milliseconds of silence at the > beginning of your sound file. :) > __ From kristian.kielhofner at gmail.com Wed Mar 11 09:32:34 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 12:32:34 -0400 Subject: [Freeswitch-users] STUN error In-Reply-To: <694251.83720.qm@web53610.mail.re2.yahoo.com> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> Message-ID: <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> On Mon, Mar 9, 2009 at 11:09 PM, Will Smith wrote: > Thank you Brian, it works like a champ. > > Yes,?west philadelfia born and raised? On a playground is where I spent most of my days... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From sprice at gmail.com Wed Mar 11 09:39:15 2009 From: sprice at gmail.com (SP) Date: Wed, 11 Mar 2009 11:39:15 -0500 Subject: [Freeswitch-users] STUN error In-Reply-To: <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> Message-ID: <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> On Wed, Mar 11, 2009 at 11:32, Kristian Kielhofner wrote: > On Mon, Mar 9, 2009 at 11:09 PM, Will Smith wrote: >> Thank you Brian, it works like a champ. >> >> Yes,?west philadelfia born and raised? > > On a playground is where I spent most of my days... Chilling out, maxing, relaxing all cool > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From mszlazak at aol.com Wed Mar 11 09:42:44 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 11 Mar 2009 12:42:44 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> Message-ID: <8CB707F89214902-624-883@webmail-da08.sysops.aol.com> I changed that tag so that hangup_after_bridge is false: ? but I still don't get the .js application working which is nothing more than a test script that ran if I dialed it's extension with the preceding one commented out: s = new Session("{ignore_early_media=true}sofia/gateway/spa3102/12223334444 at 10.0.0.5:5061"); while (s.ready()) { ??? s.answer(); ??? s.speak("cepstral","Callie","Hello World"); } -----Original Message----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Wed, 11 Mar 2009 4:34 am Subject: Re: [Freeswitch-users] Problem with "continue" in extension. The continue works fine, it just hangs up befoer that due to: ?? ? ? ? ? ? Mike On Mar 11, 2009, at 2:46 AM, mszlazak at aol.com wrote: I have the following in my dialplan. Individually, each extension does what it's suppose to do when dialing 1111. However, if I place continue=true in the first extension then it alone gets executed and the succeeding extension does not. I thought condition=true would allow the extension afterward to execute. My test hardware for this dialplan is a single PSTN line. A call comes in that line and "myExtension" executes then hopefully hangs up to free the line. Afterward, I want "myExtension_Continued" to execute the .js application and dial out that single PSTN line. I need help in getting this scenerio to work. Thanks. ?? ? ?? ??? ? ?? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ? ?? ? ?? ??? ? ?? ??? ? ?? ??? ??? ? ?? ??? ? ?? ? = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/5c898ba0/attachment-0001.html From msc at freeswitch.org Wed Mar 11 10:14:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 10:14:08 -0700 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <8CB707F89214902-624-883@webmail-da08.sysops.aol.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> <8CB707F89214902-624-883@webmail-da08.sysops.aol.com> Message-ID: <87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com> On Wed, Mar 11, 2009 at 9:42 AM, wrote: > I changed that tag so that hangup_after_bridge is false: > > ? > but I still don't get the .js application working which is nothing more than > a test script that ran if I dialed it's extension with the preceding one > commented out: Try adding a break="never" to your first extension: I believe that will cause the the dialplan to keep looking for "1111" even after it has been matched once. -MC From benke at inqnet.at Wed Mar 11 10:07:42 2009 From: benke at inqnet.at (Christian Benke) Date: Wed, 11 Mar 2009 18:07:42 +0100 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" Message-ID: <20090311180742.0c6693cb@plex> Hi! I've recently started to configure a freeswitch for our new office pbx and so far i like it very much(Coming from asterisk&openser with 2 years experience at a ITSP. Openser was nice but i didn't like asterisk for several reasons, so i searched for a more stable and cleaner alternative. Freeswitch looks _very_ promising and i'd wished i could use it for more difficult demands than a simple office-pbx ;-)). So far i had little trouble(Though our installation doesn't require much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. The only issue i have not resolved yet is setting the outgoing DID("head"-number + extension, e.g. +4312345678 + 100). The relevant part of the default.xml looks like this atm(where +4312345678 is our "head"-phone-number without the extensions, ${caller_id_number} is a 3-digit extension, e.g.: 100): I'd expect with this dialplan the effective_caller_id would be in the "From:"-section of the INVITE, but it seems after the bridge it is overwritten with the gateway-username i've defined in the gateway-configuration in sip_profiles/external/. So instead of: From: "Desk Phone" ;tag=U6yQUSta2c2Xg. i get: From: "Desk Phone" ;tag=U6yQUSta2c2Xg. in the INVITE towards the sip-trunk. I may not have grasped yet how proper debugging with freeswitch works, however, in the console the last action i see, before the bridge to sofia/external is created, is the setting of the effective-caller-id, as expected(Do you want to see the whole output?). I guess i don't necessarily need to register with the provider, as they have configured the trunk for my ip-adress and i have theirs in the ACL(inbound calls work flawless with the head-number+extension), so maybe the registration is the reason why freeswitch does that automatically? It's probably a little issue, but i don't have the overview yet to understand how this happens, maybe someone can point me to the right place? Cheers Christian From mszlazak at aol.com Wed Mar 11 10:42:29 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 11 Mar 2009 13:42:29 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com><8CB707F89214902-624-883@webmail-da08.sysops.aol.com> <87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com> Message-ID: <8CB7087E25E4398-624-C4E@webmail-da08.sysops.aol.com> Mike, no luck with that either. I still need to see this through but another related approach will be needed later so I'll ask now. Does FreeSwitch have some script or something to set up an "auto dialer." Basically, I want to be able the store some caller info then have FS automatically check to see if a "reminder" calls need to be sent out. Thanks.? -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 11 Mar 2009 10:14 am Subject: Re: [Freeswitch-users] Problem with "continue" in extension. On Wed, Mar 11, 2009 at 9:42 AM, wrote: > I changed that tag so that hangup_after_bridge is false: > > ? > but I still don't get the .js application working which is nothing more than > a test script that ran if I dialed it's extension with the preceding one > commented out: Try adding a break="never" to your first extension: I believe that will cause the the dialplan to keep looking for "1111" even after it has been matched once. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/30c04acf/attachment.html From miles.chet at gmail.com Wed Mar 11 10:48:37 2009 From: miles.chet at gmail.com (roberto) Date: Wed, 11 Mar 2009 14:48:37 -0300 Subject: [Freeswitch-users] suggestions to application In-Reply-To: <87f2f3b90903110846j69d73a0ct58a4b41f3c06168e@mail.gmail.com> References: <87f2f3b90903110846j69d73a0ct58a4b41f3c06168e@mail.gmail.com> Message-ID: thanks On Wed, Mar 11, 2009 at 12:46 PM, Michael Collins wrote: >> I?m looking for suggestion on some features like: >> >> Accounting >> Rate >> Billing >> Route > > There's nothing out-of-box ready, but you should look at these mods: > mod_nibblebill - real-time billing > mod_lcr - outbound routing > mod_easyroute - inbound routing > > Start with the wiki: http:wiki.freeswitch.org > > There's a lot of research for you to do. Have fun! :) > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- "Without love, we are birds with broken wings." Morrie From kristian.kielhofner at gmail.com Wed Mar 11 11:48:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 14:48:50 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? Message-ID: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> Hello everyone, I have an issue where FS seems to get confused in the presence of multiple Record-Route headers. SIP capture here: http://admin.star2star.com/fs-sip.log I've never seen this with FS before but it appears to process the multiple Record-Route headers backwards, at least in this case. I want to verify: 1) These Record-Route headers are syntactically correct (looks good to me). 2) FS should, in fact, process Record-Route headers "top down" and built its Route: headers (and reply) accordingly. At first I thought the FS/Sofia SIP parser may have been getting confused because the Record-Route from my proxy (.186) does not have a port in the URI. I tried adding a Record-Route header with a port - no difference. This is currently running trunk rev 12218 but I'm about to update to 12571 to see what happens. Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From msc at freeswitch.org Wed Mar 11 11:53:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 11:53:18 -0700 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <8CB7087E25E4398-624-C4E@webmail-da08.sysops.aol.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> <8CB707F89214902-624-883@webmail-da08.sysops.aol.com> <87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com> <8CB7087E25E4398-624-C4E@webmail-da08.sysops.aol.com> Message-ID: <87f2f3b90903111153h23c477fei62a4bfaccf66664f@mail.gmail.com> On Wed, Mar 11, 2009 at 10:42 AM, wrote: > Mike, no luck with that either. > You may have to get fancy and use the transfer app and create an extension like this: Then just call it from the regular 1111 extension: > I still need to see this through but another related approach will be needed > later so I'll ask now. > > Does FreeSwitch have some script or something to set up an "auto dialer." > Basically, I want to be able the store some caller info then have FS > automatically check to see if a "reminder" calls need to be sent out. > you do you have the sched_api family of API functions: http://wiki.freeswitch.org/wiki/Mod_commands#sched_api I don't have it wikified but there is also an unsched_api function so that you can cancel a future scheduled api. The API could be something like "originate user/1000 &bridge(foo/bar)" -MC From msc at freeswitch.org Wed Mar 11 11:54:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 11:54:38 -0700 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> Message-ID: <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> > ?This is currently running trunk rev 12218 but I'm about to update to > 12571 to see what happens. To quote Samuel L. Jackson in "Jurassic Park": Hold on to your butts! -MC From kristian.kielhofner at gmail.com Wed Mar 11 12:21:36 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 15:21:36 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> Message-ID: <2d9149cd0903111221p7039037qe468768b959ce3d9@mail.gmail.com> On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins wrote: >> ?This is currently running trunk rev 12218 but I'm about to update to >> 12571 to see what happens. > > To quote Samuel L. Jackson in "Jurassic Park": > Hold on to your butts! > > -MC Yeah, I know. It's just that it's an AstLinux machine and my build machine is REALLY slow... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From mszlazak at aol.com Wed Mar 11 12:26:59 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 11 Mar 2009 15:26:59 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <87f2f3b90903111153h23c477fei62a4bfaccf66664f@mail.gmail.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com><8CB707F89214902-624-883@webmail-da08.sysops.aol.com><87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com><8CB7087E25E4398-624-C4E@webmail-da08.sysops.aol.com> <87f2f3b90903111153h23c477fei62a4bfaccf66664f@mail.gmail.com> Message-ID: <8CB70967B6A20B2-B48-363@webmail-da08.sysops.aol.com> Mike, can you give me a pointer to what needs cleaning up so I have an idea how to make one of these? Also, thanks a lot for the sched_api and unsched_api, that's terrific news! -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 11 Mar 2009 11:53 am Subject: Re: [Freeswitch-users] Problem with "continue" in extension. On Wed, Mar 11, 2009 at 10:42 AM, wrote: > Mike, no luck with that either. > You may have to get fancy and use the transfer app and create an extension like this: Then just call it from the regular 1111 extension: > I still need to see this through but another related approach will be needed > later so I'll ask now. > > Does FreeSwitch have some script or something to set up an "auto dialer." > Basically, I want to be able the store some caller info then have FS > automatically check to see if a "reminder" calls need to be sent out. > you do you have the sched_api family of API functions: http://wiki.freeswitch.org/wiki/Mod_commands#sched_api I don't have it wikified but there is also an unsched_api function so that you can cancel a future scheduled api. The API could be something like "originate user/1000 &bridge(foo/bar)" -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/905a84e6/attachment-0001.html From Prometheus001 at gmx.net Wed Mar 11 14:03:27 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 11 Mar 2009 22:03:27 +0100 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app during bridge between softphone ext 1000 and PSTN. In-Reply-To: References: Message-ID: <49B8271F.7050705@gmx.net> My experience is, it can depend on the PSTN provider. I had problems with Sonus systems (see your SIP Invite message), there is a workaround in the wiki, but it didn't help in all my cases. I also have tested the start_dtmf application for additonal dtmf inband detection which helped in some cases. I also tried which didn't help in my cases + blocks routing of DTMFs through freeswitch. Best regards Peter Shahal Hazan schrieb: > > Hi, > > After I call the external number successfully, I?m able to receive > DTMF from the softphone but the PSTN?s DTMF doesn?t work. > > I am able to change the mute status during a conference from the same > PSTN. > > I had a look at: http://wiki.freeswitch.org/wiki/Dtmf_troubleshooting > but it didn?t help. > > I have: > > > > on the Local_Extension and on my SIP_PROVIDER extension, both at > sofia/default > > Any ideas? > > Shahal Hazan > > Mobile Team > Jajah > > My blog: http://jajahdevblog.com/shahal > > www.Jajah.com > > M: +972-54-227-9567 > > > > This mail was sent via Mail-SeCure System. > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From leon at scarlet-internet.nl Wed Mar 11 14:59:19 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Wed, 11 Mar 2009 22:59:19 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> Message-ID: <0DA60EA4-0873-48E9-A263-A20D14BF5A19@scarlet-internet.nl> Hi, I didn't try this new functionality yet, but shouldn't http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type be in a different section ? It's not SDP manipulation, or is it ? regards, Leon On Mar 10, 2009, at 9:48 PM, Michael Collins wrote: > On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale > wrote: >> Latest SVN: >> >> Send no extra caller id info: >> {sip_cid_type=none}sofia/default/user at example.com >> >> Send RPID (default) >> {sip_cid_type=rpid}sofia/default/user at example.com >> >> Send P-XXX-Identity >> {sip_cid_type=pid}sofia/default/user at example.com >> >> Send RPID with chosen content >> {sip_cid_type >> = >> rpid >> ,origination_caller_id_name >> =test,origination_caller_id_number=1234,origination_privacy=screen >> +hide_name+hide_number}sofia/defaul tuser at example.com > > FYI, I added this info to the channel variables page: > http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Wed Mar 11 15:19:47 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 11 Mar 2009 18:19:47 -0400 Subject: [Freeswitch-users] STUN error In-Reply-To: <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> Message-ID: <49B83903.3090306@freeswitch.org> SP wrote: > On Wed, Mar 11, 2009 at 11:32, Kristian Kielhofner > wrote: > >> On Mon, Mar 9, 2009 at 11:09 PM, Will Smith wrote: >> >>> Thank you Brian, it works like a champ. >>> >>> Yes, west philadelfia born and raised? >>> >> On a playground is where I spent most of my days... >> > > Chilling out, maxing, relaxing all cool > > OK, couldn't resist "shootin' some B-Ball outside the school" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/7381c524/attachment.html From intralanman at freeswitch.org Wed Mar 11 15:24:00 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 11 Mar 2009 18:24:00 -0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> Message-ID: <49B83A00.7000706@freeswitch.org> Anthony Minessale wrote: > try now using sip_from_display variable to control the display field > in the from url It would be awesome if someone from the community would document all of these nice new variables on the wiki. -Ray From kjv at ken-ton.com Wed Mar 11 17:19:50 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Wed, 11 Mar 2009 20:19:50 -0400 Subject: [Freeswitch-users] STUN error In-Reply-To: <49B83903.3090306@freeswitch.org> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> <49B83903.3090306@freeswitch.org> Message-ID: <1AEFE3D0-F580-40F0-81E8-BA51E4385BD5@ken-ton.com> Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-448-3009 x0 On Mar 11, 2009, at 6:19 PM, Raymond Chandler wrote: > SP wrote: >> >> On Wed, Mar 11, 2009 at 11:32, Kristian Kielhofner >> wrote: >> >>> On Mon, Mar 9, 2009 at 11:09 PM, Will Smith >>> wrote: >>> >>>> Thank you Brian, it works like a champ. >>>> >>>> Yes, west philadelfia born and raised? >>>> >>> On a playground is where I spent most of my days... >>> >> Chilling out, maxing, relaxing all cool >> >> > OK, couldn't resist > "shootin' some B-Ball outside the school" When a couple a' guys who were up to no good > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/16e2135c/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/16e2135c/attachment.bin From brian at freeswitch.org Wed Mar 11 17:28:51 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Mar 2009 19:28:51 -0500 Subject: [Freeswitch-users] STUN error In-Reply-To: <1AEFE3D0-F580-40F0-81E8-BA51E4385BD5@ken-ton.com> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> <49B83903.3090306@freeswitch.org> <1AEFE3D0-F580-40F0-81E8-BA51E4385BD5@ken-ton.com> Message-ID: <80C0FDB7-83D7-40D7-9090-C24D5022FF6C@freeswitch.org> On Mar 11, 2009, at 7:19 PM, Karl Vesterling wrote: >> OK, couldn't resist >> "shootin' some B-Ball outside the school" > When a couple a' guys who were up to no good >> Started making trouble in my neighbourhood I got in one little fight and my mom got scared And said youre moving with your aunte and uncle in bel-air /b PS: fun has been had! From kristian.kielhofner at gmail.com Wed Mar 11 19:54:46 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 22:54:46 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> Message-ID: <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins wrote: >> ?This is currently running trunk rev 12218 but I'm about to update to >> 12571 to see what happens. > > To quote Samuel L. Jackson in "Jurassic Park": > Hold on to your butts! > > -MC Just got around to trying again on 12571 - same result. Here it is again just the OK and the ACK this time: U 208.38.149.186:5060 -> 71.228.78.51:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 71.228.78.51:5060;received=71.228.78.51;rport=5060;branch=z9hG4bK4D194FH359yaH. To: ;tag=as4ba5ae74. From: "Extension 1000" ;tag=4veH7cg0XS04r. Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. CSeq: 112294452 INVITE. Content-Type: application/sdp. Contact: . Content-Length: 285. Record-Route: . User-Agent: Packetrino. Supported: replaces. Record-Route: . Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. . v=0. o=root 10960 10961 IN IP4 64.2.142.73. s=session. c=IN IP4 64.2.142.73. t=0 0. m=audio 18680 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/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 - - - -. a=ptime:20. a=sendrecv. U 71.228.78.51:5060 -> 64.2.142.93:5060 ACK sip:19412848354 at 64.2.142.73 SIP/2.0. Via: SIP/2.0/UDP 71.228.78.51;rport;branch=z9hG4bK5pt26a262jNXc. Route: . Route: . Max-Forwards: 70. From: "Extension 1000" ;tag=4veH7cg0XS04r. To: ;tag=as4ba5ae74. Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. CSeq: 112294452 ACK. Contact: . Content-Length: 0. . Trying to be as self-sufficient as I can, it looks like the code for this is on line 4268 of src/mod/endpoints/mod_sofia/sofia.c. I just wish I new what to do to it... ;) Am I the only one that has experienced this? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Wed Mar 11 20:13:13 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 23:13:13 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> Message-ID: <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> After reading into this more it looks like the Record-Route headers are to be parsed in reverse order (which FS is doing). Sorry! On Wed, Mar 11, 2009 at 10:54 PM, Kristian Kielhofner wrote: > On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins wrote: >>> ?This is currently running trunk rev 12218 but I'm about to update to >>> 12571 to see what happens. >> >> To quote Samuel L. Jackson in "Jurassic Park": >> Hold on to your butts! >> >> -MC > > Just got around to trying again on 12571 - same result. ?Here it is > again just the OK and the ACK this time: > > U 208.38.149.186:5060 -> 71.228.78.51:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > 71.228.78.51:5060;received=71.228.78.51;rport=5060;branch=z9hG4bK4D194FH359yaH. > To: ;tag=as4ba5ae74. > From: "Extension 1000" ;tag=4veH7cg0XS04r. > Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. > CSeq: 112294452 INVITE. > Content-Type: application/sdp. > Contact: . > Content-Length: 285. > Record-Route: . > User-Agent: Packetrino. > Supported: replaces. > Record-Route: . > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > . > v=0. > o=root 10960 10961 IN IP4 64.2.142.73. > s=session. > c=IN IP4 64.2.142.73. > t=0 0. > m=audio 18680 RTP/AVP 0 18 101. > a=rtpmap:0 PCMU/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 - - - -. > a=ptime:20. > a=sendrecv. > > > U 71.228.78.51:5060 -> 64.2.142.93:5060 > ACK sip:19412848354 at 64.2.142.73 SIP/2.0. > Via: SIP/2.0/UDP 71.228.78.51;rport;branch=z9hG4bK5pt26a262jNXc. > Route: . > Route: . > Max-Forwards: 70. > From: "Extension 1000" ;tag=4veH7cg0XS04r. > To: ;tag=as4ba5ae74. > Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. > CSeq: 112294452 ACK. > Contact: . > Content-Length: 0. > . > > ?Trying to be as self-sufficient as I can, it looks like the code for > this is on line 4268 of src/mod/endpoints/mod_sofia/sofia.c. ?I just > wish I new what to do to it... ;) > > ?Am I the only one that has experienced this? > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Wed Mar 11 20:18:47 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Mar 2009 22:18:47 -0500 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> Message-ID: Its ok... at the very least you question the norm! ;) /b On Mar 11, 2009, at 10:13 PM, Kristian Kielhofner wrote: > After reading into this more it looks like the Record-Route headers > are to be parsed in reverse order (which FS is doing). > > Sorry! From kristian.kielhofner at gmail.com Wed Mar 11 20:29:04 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 23:29:04 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> Message-ID: <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> Aww, thanks Brian! On Wed, Mar 11, 2009 at 11:18 PM, Brian West wrote: > Its ok... at the very least you question the norm! ;) > > /b > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Wed Mar 11 20:34:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Mar 2009 22:34:36 -0500 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> Message-ID: ;) I expect to see you at cluecon this year? /b On Mar 11, 2009, at 10:29 PM, Kristian Kielhofner wrote: > Aww, thanks Brian! > > On Wed, Mar 11, 2009 at 11:18 PM, Brian West > wrote: >> Its ok... at the very least you question the norm! ;) >> >> /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/2b265fe2/attachment.html From msc at freeswitch.org Wed Mar 11 20:50:13 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 11 Mar 2009 20:50:13 -0700 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> Message-ID: <83CD0007-CCC2-4420-95C5-8EC26D5C7FD2@freeswitch.org> On Mar 11, 2009, at 8:34 PM, Brian West wrote: > ;) I expect to see you at cluecon this year? > > /b > Notice how he threw in a compliment and an invite to CC but didn't actually address the question? ;) pretty sneaky bkw! BTW, the best way to come to CC is to get your boss to sponsor the event! :p -MC From brian at freeswitch.org Wed Mar 11 20:54:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Mar 2009 22:54:03 -0500 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <83CD0007-CCC2-4420-95C5-8EC26D5C7FD2@freeswitch.org> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> <83CD0007-CCC2-4420-95C5-8EC26D5C7FD2@freeswitch.org> Message-ID: <1C4F8A51-66B9-4C7D-B187-996521A3AEB6@freeswitch.org> I'm like bugs bunny here... sneaky wabbit! /b On Mar 11, 2009, at 10:50 PM, Michael S Collins wrote: > On Mar 11, 2009, at 8:34 PM, Brian West wrote: > >> ;) I expect to see you at cluecon this year? >> >> /b >> > Notice how he threw in a compliment and an invite to CC but didn't > actually address the question? ;) pretty sneaky bkw! > > BTW, the best way to come to CC is to get your boss to sponsor the > event! :p > > -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/d677e88f/attachment.html From wiltingtree at gmail.com Wed Mar 11 22:02:46 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Thu, 12 Mar 2009 01:02:46 -0400 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails Message-ID: Hi, I'm trying to use FreeSWITCH's mod_voicemail to replace the voicemail on wireless phones. Does anybody know how to make the wireless phone know there is a voicemail waiting, so it can notify the user?Thanks for the help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/9dea6725/attachment.html From krice at freeswitch.org Wed Mar 11 22:09:35 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 12 Mar 2009 00:09:35 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: Message-ID: When you say wireless do you mean like Cellular Phone? From: Adam Wilt Reply-To: Date: Thu, 12 Mar 2009 01:02:46 -0400 To: Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails Hi, I'm trying to use FreeSWITCH's mod_voicemail to replace the voicemail on wireless phones. Does anybody know how to make the wireless phone know there is a voicemail waiting, so it can notify the user? Thanks for the help! _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/70ad5e87/attachment.html From kawarod at laposte.net Thu Mar 12 03:15:08 2009 From: kawarod at laposte.net (rod) Date: Thu, 12 Mar 2009 14:15:08 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> Message-ID: <49B8E0AC.6070307@laposte.net> Thanks a lot Anthony, it's working great. I'm just checking the origination_privacy parameter, cause it seems to do nothing in my setup. Anthony Minessale wrote: > oops hit send too soon, > > try now using sip_from_display variable to control the display field > in the from url > > > On Wed, Mar 11, 2009 at 10:14 AM, Anthony Minessale > > wrote: > > try now > > > On Wed, Mar 11, 2009 at 9:09 AM, rod > wrote: > > ok, > > almost perfect :p > > using this in dialplan: > {sip_cid_type=rpid,origination_caller_id_number=000000${CALLER_NUMBER},origination_caller_id_name=test,originatio\ > n_privacy=screen+hide_name+hide_number,sip_from_uri=sip:anonymous at anonymous.invalid}sofia/ss7/000000${destination_number}@${GW_IP} > > I have this on B leg > From: "test" ;tag=gQ53Hjp617BSH. > Remote-Party-ID: "test" > >;party=calling;screen=yes;privacy=off. > > As you can see the caller id name is still modified, do you > think you > could add something like sip_from_uri_name to circumvent this. > > Anthony, I tried many settings for origination_privacy and it > seems to > do nothing on the RPID header. Any clue? > > regards. > > Anthony Minessale wrote: > > > > > > On Wed, Mar 11, 2009 at 3:33 AM, rod > > >> > wrote: > > > > thanks a lot Anthony, > > > > I consider the work is done and I did the paypal > transfer of 200$. > > > > But I'll be happy if you could do some minor changes: > > > > > > Done....r12563 > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shahal at jajah.com Thu Mar 12 03:41:30 2009 From: shahal at jajah.com (Shahal Hazan) Date: Thu, 12 Mar 2009 12:41:30 +0200 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 48 In-Reply-To: References: Message-ID: > After I call the external number successfully, I?m able to receive DTMF from > the softphone but the PSTN?s DTMF doesn?t work. We definitely don't want to assume anything, so I have to ask the obvious questions: who is the provider? are the DTMFs in-band or RFC2833? Any chance you can turn on full debugging and see if there are any clues? Thanks! -MC Hi, After turning on the debug in CLI I typed *3 on the PSTN and I got: 2009-03-12 11:22:49 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(0) 2009-03-12 11:22:51 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(3) Please note that I got DTMF(0) and not a DTMF(*)! I also double checked with wireshark and saw that the DTMF is SIP based, and the values were *3 and not 03 as FreeSWITCH reports. This probably is the problem and not the bind_meta_app. We are using bezeq international as our provider. DTMF are RFC2833. Thanks, Shahal This mail was sent via Mail-SeCure System. From shahal at jajah.com Thu Mar 12 03:43:33 2009 From: shahal at jajah.com (Shahal Hazan) Date: Thu, 12 Mar 2009 12:43:33 +0200 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app In-Reply-To: References: Message-ID: > After I call the external number successfully, I?m able to receive DTMF from > the softphone but the PSTN?s DTMF doesn?t work. We definitely don't want to assume anything, so I have to ask the obvious questions: who is the provider? are the DTMFs in-band or RFC2833? Any chance you can turn on full debugging and see if there are any clues? Thanks! -MC Hi, After turning on the debug in CLI I typed *3 on the PSTN and I got: 2009-03-12 11:22:49 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(0) 2009-03-12 11:22:51 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(3) Please note that I got DTMF(0) and not a DTMF(*)! I also double checked with wireshark and saw that the DTMF is SIP based, and the values were *3 and not 03 as FreeSWITCH reports. This probably is the problem and not the bind_meta_app. We are using bezeq international as our provider. DTMF are RFC2833. Thanks, Shahal This mail was sent via Mail-SeCure System. From shahal at jajah.com Thu Mar 12 03:43:53 2009 From: shahal at jajah.com (Shahal Hazan) Date: Thu, 12 Mar 2009 12:43:53 +0200 Subject: [Freeswitch-users] Recall: Freeswitch-users Digest, Vol 33, Issue 48 Message-ID: Shahal Hazan would like to recall the message, "Freeswitch-users Digest, Vol 33, Issue 48". This mail was sent via Mail-SeCure System. From william.suffill at gmail.com Thu Mar 12 05:36:27 2009 From: william.suffill at gmail.com (William Suffill) Date: Thu, 12 Mar 2009 08:36:27 -0400 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: References: Message-ID: <6b65470d0903120536m44a431aawaad1241172692358@mail.gmail.com> If they aren't attached directly to the FS box I guess you could call them and play a recording and allow them to access their VM by creating an outbound call w/ an ivr to a) see if u got the person vs VM etc then b) allow them to get to their VM. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/4c29c29d/attachment.html From yudha2008 at gmail.com Thu Mar 12 06:47:44 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 12 Mar 2009 19:17:44 +0530 Subject: [Freeswitch-users] openzap with A102 Message-ID: Hi, I am using sangoma A102 card with freeswitch. I have updated all the changes in the freeswitch and have loaded openzap also. But still i cant able to make an outbound call. *openzap.conf* [span wanpipe] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 [span wanpipe] name => OpenZAP number => 2 trunk_type => E1 b-channel => 2:1-15 d-channel => 2:16 b-channel => 2:17-31 *openzap.conf.xml* *default.xml * - * I have attached the freeswitch log http://pastebin.freeswitch.org/7730 Can any one correct were i am wrong. * -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/95cc8b7b/attachment.html From kawarod at laposte.net Thu Mar 12 07:12:04 2009 From: kawarod at laposte.net (rod) Date: Thu, 12 Mar 2009 18:12:04 +0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value Message-ID: <49B91834.6050004@laposte.net> Hi list, I'm testing mod_limit like this: where AREA could be any value like: US/EUROPE/AFRICA... that has been set for the call then I use sipp for load testing with 4 cps and max 65 calls so that I will be limited by the limit of 60 in the dialplan. I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit value growing up to 60. Sipp still tries to establish new call (up to 65 calls) at 4cps and for each new cps in excess, FS sends a 503. I wait for 10 seconds and stop sipp, but the limit value is never decreasing even when there is no more channels used (show channels), the limit value is stuck to 60. If I limit Sipp to 55 calls (below the limit value), the limit value increase and decrease depending on load, and the pbm doesn't appear. Does anybody is using mod_limit and have encountered the same pbm. I'm using latest svn: 12580. regards, rod From mrene_lists at avgs.ca Thu Mar 12 08:04:24 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 12 Mar 2009 11:04:24 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49B91834.6050004@laposte.net> References: <49B91834.6050004@laposte.net> Message-ID: <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> Were you doing transfers (action="transfer" in the dialplan) ? If yes, retry with revision 12581, and open a JIRA if it still is an issue (also include a full debug log (delete your freeswitch.log file before doing your test and attach it after) in your JIRA) Math On 12-Mar-09, at 10:12 AM, rod wrote: > Hi list, > > I'm testing mod_limit like this: > > > > > where AREA could be any value like: US/EUROPE/AFRICA... that has been > set for the call > > then I use sipp for load testing with 4 cps and max 65 calls so that I > will be limited by the limit of 60 in the dialplan. > > I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit > value > growing up to 60. > > Sipp still tries to establish new call (up to 65 calls) at 4cps and > for > each new cps in excess, FS sends a 503. > I wait for 10 seconds and stop sipp, but the limit value is never > decreasing even when there is no more channels used (show channels), > the > limit value is stuck to 60. > > > If I limit Sipp to 55 calls (below the limit value), the limit value > increase and decrease depending on load, and the pbm doesn't appear. > > Does anybody is using mod_limit and have encountered the same pbm. > > I'm using latest svn: 12580. > > regards, > rod > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Mark.Tabron at rnid-typetalk.org.uk Thu Mar 12 06:54:49 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Thu, 12 Mar 2009 13:54:49 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 Message-ID: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> Hi, My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn't quite correct. Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel. The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out) Here are the confs / dialplan as I have them: [span wanpipe] name => OpenZAP number => 1 trunk_type => e1 b-channel => 1:1-9 d-channel => 1:16 Do the config files look ok? Is there anything else I should be configuring? Is there anything else I can use to debug or get information on the PRI status? Thanks in advance. Mark. Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/b111162a/attachment-0001.html From ax.russo at gmail.com Thu Mar 12 08:01:46 2009 From: ax.russo at gmail.com (Alessandro Russo) Date: Thu, 12 Mar 2009 16:01:46 +0100 Subject: [Freeswitch-users] Freeswitch and OPAL/H323 Message-ID: Hi to all, I am a newbie in Freeswitch (FS). I have already installed a FS machine, following the wiki installation procedure, and I have also added the opal module following this procedure: http://wiki.freeswitch.org/wiki/Mod_opal When I running FS ###################################### freeswitch at atest> module_exists mod_opal API CALL [module_exists(mod_opal)] output: true freeswitch at atest> ###################################### I think I'have installed it correctly. My goal is to provide a conference tool for incoming h323-calls that come from a cisco call manager: we are behind a Cisco VoIP cloud. Every time a PSTN phone calls the number 1234-123456 the call manager knows that the extension 3456 has to be redirected to 192.168.193.38, that is the IP address of the FS machine, where file dialplan/default/myconference.xml contains the following lines ###################################### ###################################### On the other hand, whenever a user of FS calls an local extension (like 1XXX ), what I want is that FS forward this call to the cisco call manager through opal/h323 therefore I have a file in ###################################### ###################################### but it fails: when a FS user calls 1500, FS returns this message ###################################### 2009-03-12 15:55:08 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1000 at 192.168.193.38[c7b69402-0f15-11de-b4dc-c11b39fce37c] 2009-03-12 15:55:08 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1500 in context default 2009-03-12 15:55:08 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel opal/h323:1500 at IP.CALL.MANA.GER:1720 [c7c010e0-0f15-11de-b4dc-c11b39fce37c] 2009-03-12 15:55:08 [INFO] h323pdu.cxx:999 H225() Read error (0): 2009-03-12 15:55:08 [NOTICE] mod_opal.cpp:591 OnReleased() Hangup opal/h323:1500 at IP.CALL.MANA.GER:1720 [CS_CONSUME_MEDIA] [UNKNOWN] 2009-03-12 15:55:08 [INFO] tlibthrd.cxx:363 PWLib() Destroyed thread 0xb171a708 H225 Caller:0xa9587b90(id = 0) 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 22 (opal/h323:1500 at IP.CALL.MANA.GER:1720) Ended 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel opal/h323:1500 at IP.CALL.MANA.GER:1720 [CS_HANGUP] 2009-03-12 15:55:08 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: UNKNOWN 2009-03-12 15:55:08 [NOTICE] mod_dptools.c:596 hangup_function() Hangup sofia/internal/1000 at 192.168.193.38 [CS_EXECUTE] [NORMAL_CLEARING] 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 21 (sofia/internal/1000 at 192.168.193.38) Ended 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/ 1000 at 192.168.193.38 [CS_HANGUP] ###################################### I don't understand why.... Any suggestions... Thanks Alessandro R. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/09216e6d/attachment.html From cstomi.levlist at gmail.com Thu Mar 12 09:06:57 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Thu, 12 Mar 2009 17:06:57 +0100 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> Message-ID: <49B93321.5080500@gmail.com> Hello, we had the same problem. we couldn't test r12581 for sure yet, but we will. This fix is only for the limit (db version), right? Would be limit_hash a better choice to increase performance anyway? Rod, do yoo have maybe experiences with limit_hash with your sipp? Thanks in advance, Tamas Mathieu Rene ?rta: > Were you doing transfers (action="transfer" in the dialplan) ? > > If yes, retry with revision 12581, and open a JIRA if it still is an > issue (also include a full debug log (delete your freeswitch.log file > before doing your test and attach it after) in your JIRA) > > Math > > On 12-Mar-09, at 10:12 AM, rod wrote: > > >> Hi list, >> >> I'm testing mod_limit like this: >> >> >> >> >> where AREA could be any value like: US/EUROPE/AFRICA... that has been >> set for the call >> >> then I use sipp for load testing with 4 cps and max 65 calls so that I >> will be limited by the limit of 60 in the dialplan. >> >> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >> value >> growing up to 60. >> >> Sipp still tries to establish new call (up to 65 calls) at 4cps and >> for >> each new cps in excess, FS sends a 503. >> I wait for 10 seconds and stop sipp, but the limit value is never >> decreasing even when there is no more channels used (show channels), >> the >> limit value is stuck to 60. >> >> >> If I limit Sipp to 55 calls (below the limit value), the limit value >> increase and decrease depending on load, and the pbm doesn't appear. >> >> Does anybody is using mod_limit and have encountered the same pbm. >> >> I'm using latest svn: 12580. >> >> regards, >> rod >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu Mar 12 09:50:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Mar 2009 09:50:07 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> > My first post to the list. I?m a bit of a newb to FreeSwitch (and linux) so > apologies if some of my terminology isn?t quite correct. Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > > > > Recently had a 9 channel ISDN30 (euro ? q931) installed by BT (UK). We?ve > hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the > card is green and wanrouter is installed and up in TDM_API mode, with the > connection status showing as connected. ?Configured Openzap for 9 b and 1 d > channel as described in Freeswitch Wiki. Then created a diaplan to fire off > any calls preceded by 9 to the next available openzap channel. Looks good so far... > The problem I have is when I initiate an external call (using 9xxxxxxx) from > an extension I can see Freeswitch allocating the call to the next available > channel but then the just sits there and times out after 1 minute. With the > cause stated as ORIGINATOR_CANCEL (guessing this is the time out) okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has lots of useful information for how to gather log information, how to use the pastebin, etc. Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers: To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org). Welcome to the wonderful world of telephony debugging! -MC P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.) From msc at freeswitch.org Thu Mar 12 09:57:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Mar 2009 09:57:38 -0700 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: References: Message-ID: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> > I have attached the freeswitch log http://pastebin.freeswitch.org/7730 > > Can any one correct were i am wrong. Baskar, At first look your configs seem okay. Please pastebin the output of these commands from the CLI: oz list oz dump 1 -MC From msc at freeswitch.org Thu Mar 12 10:04:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Mar 2009 10:04:53 -0700 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app In-Reply-To: References: Message-ID: <87f2f3b90903121004y1e4bc35epe3eef28da0993a65@mail.gmail.com> > I also double checked with wireshark and saw that the DTMF is SIP based, and the values were *3 and not 03 as FreeSWITCH reports. > This probably is the problem and not the bind_meta_app. > We are using bezeq international as our provider. > DTMF are RFC2833. > Thanks, > Shahal Shahal, The devs have very recently made some improvements. Can you check out the latest SVN and try this again? Let us know what happens. -MC From brian at freeswitch.org Thu Mar 12 10:13:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Mar 2009 12:13:22 -0500 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app In-Reply-To: References: Message-ID: <63832FB2-689F-4B8A-8896-ABCABD45E856@freeswitch.org> What SVN Rev are you on? I recall fixing this ... /b On Mar 12, 2009, at 5:43 AM, Shahal Hazan wrote: >> After I call the external number successfully, I?m able to receive >> DTMF from >> the softphone but the PSTN?s DTMF doesn?t work. > > We definitely don't want to assume anything, so I have to ask the > obvious questions: > who is the provider? > are the DTMFs in-band or RFC2833? > > Any chance you can turn on full debugging and see if there are any > clues? > > Thanks! > -MC > > Hi, > After turning on the debug in CLI I typed *3 on the PSTN and I got: > 2009-03-12 11:22:49 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() > INFO DTMF(0) > 2009-03-12 11:22:51 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() > INFO DTMF(3) > Please note that I got DTMF(0) and not a DTMF(*)! > I also double checked with wireshark and saw that the DTMF is SIP > based, and the values were *3 and not 03 as FreeSWITCH reports. > This probably is the problem and not the bind_meta_app. > We are using bezeq international as our provider. > DTMF are RFC2833. > Thanks, > Shahal > > > > This mail was sent via Mail-SeCure System. From e.schmidbauer at gmail.com Thu Mar 12 11:38:33 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Thu, 12 Mar 2009 14:38:33 -0400 Subject: [Freeswitch-users] RPC and web admin panel for conference? In-Reply-To: References: Message-ID: <2cef777b0903121138p1c55c7h8b4eb71427d0b901@mail.gmail.com> I would like to try out and possibly contribute to your web admin panel. I was wondering if i need flash 8 or Flash MX 2004 for this to work though. Please let me know because I do not have a copy of that software and I am not sure if I can run it on linux (which I am running freeswitch on). On Tue, Mar 10, 2009 at 6:32 PM, Ken Rice wrote: > Hey, I just implemented something like this and commited it to my contrib > directory (scripts/contrib/swk ) its a mixture of amf-php, ESL, and Flex... > Its not complete by anymeans and you need Flex3 to compile the UI... > > Anyone wanting to throw some patches at it for other functionality are > welcome to do so... One thing is severly lacks at this time is ANY sort of > authentication...so you wouldn?t want it publically open to the world > > K > > > ________________________________ > From: Harry FSwitch > Reply-To: > Date: Mon, 9 Mar 2009 18:17:41 -0400 > To: > Subject: [Freeswitch-users] RPC and web admin panel for conference? > > Hi all, > > I'm looking to implement an admin panel much like the one used at > http://conference.freeswitch.org. Now I obviously cannot login and see the > "admin" part of the panel but I'm pretty sure whats in there. > > I have xml_rpc running and can connect via http and issue commands. I've > searched the forum here and went through the wiki, found nothing that looked > like a panel. I was hoping to find a panel I can just configure and > implement. Does anyone have a php (I guess, seeing as I have a php server) > panel they can share with me? I'm sure I can get it working for my system. > The thought of attempting one on my own at THIS point seems daunting at > best. > > Any help would be greatly appreciated! > > Thanks > > ________________________________ > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From switchserver at gmail.com Thu Mar 12 11:55:10 2009 From: switchserver at gmail.com (Harry FSwitch) Date: Thu, 12 Mar 2009 14:55:10 -0400 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! Message-ID: Greetings, Last week I submitted this post to the mailing list... http://www.nabble.com/Please-end-the-torment-td22352222.html I received a mixed response to say the least, no one however emailed me saying they wanted to be involved or would contribute hosting resources. I figured eh screw-it, and went ahead and created the forums anyway and opened them today! http://freeswitch411.info I went on IRC to let folks know its available and after a few jabs from the crowd Anthony said: "#freeswitch 2009-03-12 10:47:04 [anthm] harr, if you maintain it you are welcome to have it" If it takes off and provides a friendly, helpful entryway for new FreeSWITCH users then I will be happy, if it flops I'll be said. But I will maintain it and do what I can to help FreeSWITCH grow. This is the only time I'll overtly mention it on this list, I'll have the link in my signature and thats about it. :) Thanks for your attention -- Harry http://freeswitch411.info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/f8025f5e/attachment.html From intralanman at freeswitch.org Thu Mar 12 12:00:44 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 12 Mar 2009 15:00:44 -0400 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! In-Reply-To: References: Message-ID: <49B95BDC.9030407@freeswitch.org> Harry FSwitch wrote: > > If it takes off and provides a friendly, helpful entryway for new > FreeSWITCH users then I will be happy, if it flops I'll be said. But I > will maintain it and do what I can to help FreeSWITCH grow. This is > the only time I'll overtly mention it on this list, I'll have the link > in my signature and thats about it. :) > something like the following? -------------- unofficial forums http://freeswitch411.info From krice at suspicious.org Thu Mar 12 12:20:21 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 12 Mar 2009 14:20:21 -0500 Subject: [Freeswitch-users] RPC and web admin panel for conference? In-Reply-To: <2cef777b0903121138p1c55c7h8b4eb71427d0b901@mail.gmail.com> Message-ID: Its a 2 part solution Part 1 is PHP + AMF-PHP + ESL AMF-PHP is a class for PHP to do the Flash/Flex Serialized Binary Data format for communication to the back end ESL is FS's Event Socket Library that has been wrapped for use in FreeSwitch The Flex Client requires at least Flash 9 runtime to run... To compile it, you'll need either the Flex IDE, the Eclipse Flex plugin (both from adobe) or the opensource flex compiler In its current form its not meant to be a 'user facing' controller, but it can be used as a starting point. I have tried to be generic as possible about what I am implementing. K > From: e schmidbauer > Reply-To: > Date: Thu, 12 Mar 2009 14:38:33 -0400 > To: > Subject: Re: [Freeswitch-users] RPC and web admin panel for conference? > > I would like to try out and possibly contribute to your web admin > panel. I was wondering if i need flash 8 or Flash MX 2004 for this to > work though. Please let me know because I do not have a copy of that > software and I am not sure if I can run it on linux (which I am > running freeswitch on). > > On Tue, Mar 10, 2009 at 6:32 PM, Ken Rice wrote: >> Hey, I just implemented something like this and commited it to my contrib >> directory (scripts/contrib/swk ) its a mixture of amf-php, ESL, and Flex... >> Its not complete by anymeans and you need Flex3 to compile the UI... >> >> Anyone wanting to throw some patches at it for other functionality are >> welcome to do so... One thing is severly lacks at this time is ANY sort of >> authentication...so you wouldn?t want it publically open to the world >> >> K >> >> >> ________________________________ >> From: Harry FSwitch >> Reply-To: >> Date: Mon, 9 Mar 2009 18:17:41 -0400 >> To: >> Subject: [Freeswitch-users] RPC and web admin panel for conference? >> >> Hi all, >> >> I'm looking to implement an admin panel much like the one used at >> http://conference.freeswitch.org. Now I obviously cannot login and see the >> "admin" part of the panel but I'm pretty sure whats in there. >> >> I have xml_rpc running and can connect via http and issue commands. I've >> searched the forum here and went through the wiki, found nothing that looked >> like a panel. I was hoping to find a panel I can just configure and >> implement. Does anyone have a php (I guess, seeing as I have a php server) >> panel they can share with me? I'm sure I can get it working for my system. >> The thought of attempting one on my own at THIS point seems daunting at >> best. >> >> Any help would be greatly appreciated! >> >> Thanks >> >> ________________________________ >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From itche at bridgeport.edu Thu Mar 12 12:26:49 2009 From: itche at bridgeport.edu (itche at bridgeport.edu) Date: Thu, 12 Mar 2009 15:26:49 -0400 (EDT) Subject: [Freeswitch-users] javascript debug Message-ID: <3306.69.120.166.49.1236886009.squirrel@webmail.bridgeport.edu> Hello all, I'm new to this very nice system I'm looking into writing javascripts to interact with the system. How can one debug, run step by step and get variables values the javascripts running under this system? For example I have setup the sample script that do the "tone tests" but I cant find a way to single step it and to look into it. Thanks Itche From msc at freeswitch.org Thu Mar 12 12:59:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Mar 2009 12:59:13 -0700 Subject: [Freeswitch-users] javascript debug In-Reply-To: <3306.69.120.166.49.1236886009.squirrel@webmail.bridgeport.edu> References: <3306.69.120.166.49.1236886009.squirrel@webmail.bridgeport.edu> Message-ID: <87f2f3b90903121259j767ed323la52035b93e9f20d7@mail.gmail.com> On Thu, Mar 12, 2009 at 12:26 PM, wrote: > Hello all, > I'm new to this very nice system > I'm looking into writing javascripts to interact with the system. > How can one debug, run step by step and get variables values the > javascripts running under this system? For example I have setup the sample > script that do the "tone tests" but I cant find a way to single step it > and to look into it. I think you're unable to step through scripts that are launched "inside" of FS. Your best bet is to use lots of console log messages. I prefer to use the INFO level so that I don't see all the debug messages flying by: console_log("info", "This line should appear in green letters on the FS CLI.\n"); You can also print the values of variables using this function. -MC From sprice at gmail.com Thu Mar 12 13:07:44 2009 From: sprice at gmail.com (SP) Date: Thu, 12 Mar 2009 15:07:44 -0500 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! In-Reply-To: References: Message-ID: <7e2ac3270903121307x790cc236k56792f5c655702e8@mail.gmail.com> Another jab, in fun: hey look, that first link looks like a forum?? wow :D On Thu, Mar 12, 2009 at 13:55, Harry FSwitch wrote: > Greetings, > > Last week I submitted this post to the mailing list... > > http://www.nabble.com/Please-end-the-torment-td22352222.html > -- Shannon From mszlazak at aol.com Thu Mar 12 14:22:28 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 12 Mar 2009 17:22:28 -0400 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! In-Reply-To: References: Message-ID: <8CB716FC809DF3B-8FC-1005@webmail-mh34.sysops.aol.com> Great, I like forums better than lists. Make sure the folks at FreeSwitch make it know to anyone coming to any of their pages by providing links, etc. Thanks -----Original Message----- From: Harry FSwitch To: freeswitch-users at lists.freeswitch.org Sent: Thu, 12 Mar 2009 11:55 am Subject: [Freeswitch-users] FreeSWITCH forum community opened today! Greetings, Last week I submitted this post to the mailing list... http://www.nabble.com/Please-end-the-torment-td22352222.html I received a mixed response to say the least, no one however emailed me saying they wanted to be involved or would contribute hosting resources. I figured eh screw-it, and went ahead and created the forums anyway and opened them today! http://freeswitch411.info I went on IRC to let folks know its available and after a few jabs from the crowd Anthony said: "#freeswitch 2009-03-12 10:47:04 [anthm] harr, if you maintain it you are welcome to have it" If it takes off and provides a friendly, helpful entryway for new FreeSWITCH users then I will be happy, if it flops I'll be said. But I will maintain it and do what I can to help FreeSWITCH grow. This is the only time I'll overtly mention it on this list, I'll have the link in my signature and thats about it. :) Thanks for your attention -- Harry http://freeswitch411.info _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/1c17c320/attachment.html From qulix at mail.ru Thu Mar 12 14:26:52 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Fri, 13 Mar 2009 00:26:52 +0300 Subject: [Freeswitch-users] Freeswitch wav-files playback. Message-ID: Greetings People! I am starting user of freeswitch and that is my first time I met such problem. My installed pack (FreeSWITCH Version 1.0.trunk (12573).) doesnt play any wav files. That is what I recieve, when Im trying to playback 2009-03-13 00:14:26 [ERR] switch_core_file.c:71 switch_core_perform_file_open() Invalid file format [wav] for [/usr/local/freeswitch/sounds/en/us/callie/1.wav]! Ofcourse the file exists, and I played it on another work freeswitch station - and it sound. _____ at _______:/usr/local/freeswitch/sounds/en/us/callie# file 1.wav 1.wav: RIFF (little-endian) data, WAVE audio, IMA ADPCM, mono 16000 Hz My OS is Ubuntu 8.04 server. PS. One thing - no metter does wav file exist or not - the error always the same. =\ Tryied more files - same result. I also tryied to reconfigure/remake it - doesnt help. Maybe getting new trunk will help? Thanks. From brian at freeswitch.org Thu Mar 12 14:43:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Mar 2009 16:43:35 -0500 Subject: [Freeswitch-users] Freeswitch wav-files playback. In-Reply-To: References: Message-ID: <522EF23F-D3C1-433B-B887-8105851A1234@freeswitch.org> Sounds like you don' t have mod_sndfile loaded. /b On Mar 12, 2009, at 4:26 PM, ????... wrote: > _____ at _______:/usr/local/freeswitch/sounds/en/us/callie# file 1.wav > 1.wav: RIFF (little-endian) data, WAVE audio, IMA ADPCM, mono 16000 Hz From wiltingtree at gmail.com Thu Mar 12 15:39:27 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Thu, 12 Mar 2009 18:39:27 -0400 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails Message-ID: Sorry, yes I mean cellular phones. Is there some way to notify them of new voicemail messages? > > Message: 7 > Date: Thu, 12 Mar 2009 00:09:35 -0500 > From: Ken Rice > Subject: Re: [Freeswitch-users] How to notify wireless phones about > unread voicemails > To: > Message-ID: > > > Content-Type: text/plain; charset="us-ascii" > > When you say wireless do you mean like Cellular Phone? > > > > From: Adam Wilt > Reply-To: > Date: Thu, 12 Mar 2009 01:02:46 -0400 > To: > Subject: [Freeswitch-users] How to notify wireless phones about unread > voicemails > > Hi, I'm trying to use FreeSWITCH's mod_voicemail to replace the voicemail > on > wireless phones. Does anybody know how to make the wireless phone know > there > is a voicemail waiting, so it can notify the user? > Thanks for the help! > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/4b5624bc/attachment.html From rupa at rupa.com Thu Mar 12 15:45:00 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 12 Mar 2009 17:45:00 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: References: Message-ID: Send them an SMS? On Thu, Mar 12, 2009 at 5:39 PM, Adam Wilt wrote: > Sorry, yes I mean cellular phones. Is there some way to notify them of new > voicemail messages? > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/50431e20/attachment.html From brian at freeswitch.org Thu Mar 12 15:48:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Mar 2009 17:48:52 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: References: Message-ID: <22A9E1FD-232E-4152-9441-EF8E98D04B22@freeswitch.org> Yes there is a way and If I could recall exactly how.. its binary data in the SMS... but some providers IE AT&T block binary sms... so its pointless.. works with T-Mobile... You can toggle the vm, email, fax and other flags on and off via SMS. /b On Mar 12, 2009, at 5:45 PM, Rupa Schomaker wrote: > Send them an SMS? > > On Thu, Mar 12, 2009 at 5:39 PM, Adam Wilt > wrote: > Sorry, yes I mean cellular phones. Is there some way to notify them > of new voicemail messages? > > > > -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/48476cee/attachment.html From krice at freeswitch.org Thu Mar 12 16:10:26 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 12 Mar 2009 18:10:26 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: Message-ID: Theres the SMS method that other people have mention... There are other methods if you are the Cellular Provider for say a CoOp or a rural ilec... These include (but not limited too and depend on your switch) SS7 TCAP MWI message, sending the correct DTMF down the loop to the switch, etc etc... That being said its posible to use freeswitch for this... It just depends on exactly what type of interconnect you are doing From: Adam Wilt Reply-To: Date: Thu, 12 Mar 2009 18:39:27 -0400 To: Subject: Re: [Freeswitch-users] How to notify wireless phones about unread voicemails Sorry, yes I mean cellular phones. Is there some way to notify them of new voicemail messages? ? > > Message: 7 > Date: Thu, 12 Mar 2009 00:09:35 -0500 > From: Ken Rice > Subject: Re: [Freeswitch-users] How to notify wireless phones about > ? ? ? ?unread voicemails > To: > Message-ID: > > Content-Type: text/plain; charset="us-ascii" > > When you say wireless do you mean like Cellular Phone? > > > > From: Adam Wilt > Reply-To: > Date: Thu, 12 Mar 2009 01:02:46 -0400 > To: > Subject: [Freeswitch-users] How to notify wireless phones about unread > voicemails > > Hi, I'm trying to use FreeSWITCH's mod_voicemail to replace the voicemail on > wireless phones. Does anybody know how to make the wireless phone know there > is a voicemail waiting, so it can notify the user? > Thanks for the help! > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/e9a69882/attachment.html From yudha2008 at gmail.com Thu Mar 12 21:58:34 2009 From: yudha2008 at gmail.com (Baskar) Date: Fri, 13 Mar 2009 10:28:34 +0530 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> References: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> Message-ID: *Hi Michael Collins, When i load mod_openzap i can able to get this output **Successfully Loaded [mod_openzap**] with one error message 2009-03-13 10:21:42 [ERR] mod_openzap.c:1898 load_config() Error starting OpenZAP span 1 mode: -1264601207 dialect: -1264601162 error: I have pasted full output in this path : http://pastebin.freeswitch.org/7746 For oz list command i get this output.* * oz list API CALL [oz(list)] output: For oz dump 1 command i get this output: oz dump 1 API CALL [oz(dump 1)] output: -ERR invalid span Correct where i am worng. ** -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/c4603bc4/attachment.html From asannucci at gmail.com Thu Mar 12 22:33:45 2009 From: asannucci at gmail.com (Andrea) Date: Fri, 13 Mar 2009 00:33:45 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unreadvoicemails References: <22A9E1FD-232E-4152-9441-EF8E98D04B22@freeswitch.org> Message-ID: <9FF857FB3FC345918FD05DD8B756B1EA@quos> With some nokia models (with SIP client) the voicemail notify work fine. I tried it with FS with a Nokia 6300i - Andrea - -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/187d13c5/attachment.html From mszlazak at aol.com Thu Mar 12 23:38:53 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 13 Mar 2009 02:38:53 -0400 Subject: [Freeswitch-users] How do I notify FreeSwitch that a phone has been answered to play audio or TTS Message-ID: <8CB71BD82A0E313-CF4-158F@MBLK-M31.sysops.aol.com> If I originate an outgoing call from FreeSwitch and want to tts a phrase or play an audio once the call has been answered (i.e, someone answered their cell phone or I got their voicemail) then how do I detect that. Otherwise, I've tried the following but it relies in getting the timing right which won't always work or looping the tts phrase over and over. var s; while (tryCalling()) {} s.hangup(); exit(); function tryCalling() { ??? s = new Session("sofia/gateway/spa3102/12223334444 at 10.0.0.5:5061"); ??? s.waitForAnswer(10000); ??? ??? if (s.cause == "USER_BUSY") { ??? ?? ?return true; ??? } ??? if (s.ready()) { ??? ?? ?s.sleep(10000); ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); ??? } ??? return false; } Another way is to keep replaying the tts phrase by replacing ? if (s.ready()) { ??? ?? ?s.sleep(10000); ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); ??? } with something like: while (s.ready()) { ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); ??? } Thanks. Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/4574725b/attachment-0001.html From jalsot at gmail.com Fri Mar 13 01:46:39 2009 From: jalsot at gmail.com (Tamas) Date: Fri, 13 Mar 2009 09:46:39 +0100 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: References: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> Message-ID: <49BA1D6F.30002@gmail.com> Hello, please show your configs (openzap.conf, openzap.conf.xml). Which protocol do you intend to use? (EuroISDN, etc.) Regards, Tamas Baskar ?rta: > *Hi Michael Collins, > > When i load mod_openzap i can able to get this output > **Successfully Loaded [mod_openzap**] with one error message > > 2009-03-13 10:21:42 [ERR] mod_openzap.c:1898 load_config() Error > starting OpenZAP span 1 mode: -1264601207 dialect: -1264601162 error: > > I have pasted full output in this path : > http://pastebin.freeswitch.org/7746 > > For oz list command i get this output.* > > * oz list > API CALL [oz(list)] output: > > For oz dump 1 command i get this output: > > oz dump 1 > API CALL [oz(dump 1)] output: > -ERR invalid span > > Correct where i am worng. > ** > -- > Warm Regards, > N.Baskar > * > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kawarod at laposte.net Fri Mar 13 01:51:49 2009 From: kawarod at laposte.net (rod) Date: Fri, 13 Mar 2009 12:51:49 +0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49B93321.5080500@gmail.com> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> Message-ID: <49BA1EA5.4050201@laposte.net> Hello, I'm now running r12590, the pbm was still there, but this was because of a broken dialplan. I'm using this for exceeded limit: but this extension was at the end of my dialplan and I matched an other extension before reaching the limit extension. What was odd was that in ngrep I saw FS sending a 503, I will investigate on this. I will rerun long term test and let you know if all is ok, so I have to do for a previous mail related to ghost sessions in CLI. @Tamas: I just give a try to limit_hash, and didn't make many tests with it. It's on my todo list. Limit_hash is not a better choice than limit, their usage are different: - limit_hash is a good way to rate limit a specific gateway for example so that your switch won't be flooded by a misconfigured peer gw - limit is for limiting concurrent call to an extension/gateway, eg you have a peer that provides you with 30channels and you want to allow 15 channels for mobile (PLMN) and 15 for PSTN without relying If you still have pbm with limit and svn, pay attention to you dialplan :p @Mathieu I hope you didn't work on a virtual pbm, cause it seems to be a dialplan misconfiguration. I'll let you know if I still have pbm. Thanks for your help. regards. rod Tamas Cseke wrote: > Hello, > > we had the same problem. we couldn't test r12581 for sure yet, but we will. > This fix is only for the limit (db version), right? > Would be limit_hash a better choice to increase performance anyway? > Rod, do yoo have maybe experiences with limit_hash with your sipp? > > Thanks in advance, > Tamas > > > Mathieu Rene ?rta: > >> Were you doing transfers (action="transfer" in the dialplan) ? >> >> If yes, retry with revision 12581, and open a JIRA if it still is an >> issue (also include a full debug log (delete your freeswitch.log file >> before doing your test and attach it after) in your JIRA) >> >> Math >> >> On 12-Mar-09, at 10:12 AM, rod wrote: >> >> >> >>> Hi list, >>> >>> I'm testing mod_limit like this: >>> >>> >>> >>> >>> where AREA could be any value like: US/EUROPE/AFRICA... that has been >>> set for the call >>> >>> then I use sipp for load testing with 4 cps and max 65 calls so that I >>> will be limited by the limit of 60 in the dialplan. >>> >>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>> value >>> growing up to 60. >>> >>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>> for >>> each new cps in excess, FS sends a 503. >>> I wait for 10 seconds and stop sipp, but the limit value is never >>> decreasing even when there is no more channels used (show channels), >>> the >>> limit value is stuck to 60. >>> >>> >>> If I limit Sipp to 55 calls (below the limit value), the limit value >>> increase and decrease depending on load, and the pbm doesn't appear. >>> >>> Does anybody is using mod_limit and have encountered the same pbm. >>> >>> I'm using latest svn: 12580. >>> >>> regards, >>> rod >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From Richard.Lamkin at mettoni.com Fri Mar 13 03:09:58 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Fri, 13 Mar 2009 10:09:58 -0000 Subject: [Freeswitch-users] Sending SUBCRIBE request to a peer PBX Message-ID: <3181A30B8C35AB4AA8577B78DDF461380499DBC8@nickel.mettonigroup.com> Dear All, I have an external gateway defined which registers as a client to another SIP PBX [a Nortel CS2K]. In effect FS is an extension of the CS2K. The CS2K can provide presence info of the other extensions using the SUBSCRIBE method. Is there a way to issue SUBSCIBE and handle [record or forward as an event] the returned NOTIFY. An example would be using Xlite as the client and it can harvest the presence info for softphones in the contact list. I have tested that FS delivers presence info to clients of it. Regards Richard Lamkin ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/529b97ca/attachment.html From regs at kinetix.gr Fri Mar 13 06:25:08 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Fri, 13 Mar 2009 15:25:08 +0200 Subject: [Freeswitch-users] Call-Direction issue Message-ID: <49BA5EB4.7040907@kinetix.gr> Hi, I am trying to include the Call-Direction variable (${direction}) in my csv cdrs. But I get nothing. I tried printing the variable in the console before a bridge : But nothing there too. When I use uuid_dump during a call I can see the Call-Direction variable. I tried call_direction as well but it doesn't work either. Any ideas? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From Richard.Lamkin at mettoni.com Fri Mar 13 06:29:25 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Fri, 13 Mar 2009 13:29:25 -0000 Subject: [Freeswitch-users] Load testing on XP Message-ID: <3181A30B8C35AB4AA8577B78DDF461380499DD68@nickel.mettonigroup.com> Dear All, Does anyone have any load test results using XP(SP3) - I have checked the wiki but could not find any recorded. Regards Richard Lamkin ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/f2911933/attachment-0001.html From mike at jerris.com Fri Mar 13 06:33:20 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 13 Mar 2009 09:33:20 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA1EA5.4050201@laposte.net> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> Message-ID: <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> What is a pbm? On Mar 13, 2009, at 4:51 AM, rod wrote: > Hello, > > I'm now running r12590, the pbm was still there, but this was > because of > a broken dialplan. > > I'm using this for exceeded limit: > > expression="^limit_exceeded$"> > > > > > > but this extension was at the end of my dialplan and I matched an > other > extension before reaching the limit extension. > What was odd was that in ngrep I saw FS sending a 503, I will > investigate on this. > > I will rerun long term test and let you know if all is ok, so I have > to > do for a previous mail related to ghost sessions in CLI. > > @Tamas: > I just give a try to limit_hash, and didn't make many tests with it. > It's on my todo list. Limit_hash is not a better choice than limit, > their usage are different: > - limit_hash is a good way to rate limit a specific gateway for > example so that your switch won't be flooded by a misconfigured peer > gw > - limit is for limiting concurrent call to an extension/gateway, eg > you have a peer that provides you with 30channels and you want to > allow > 15 channels for mobile (PLMN) and 15 for PSTN without relying > > If you still have pbm with limit and svn, pay attention to you > dialplan :p > > @Mathieu > I hope you didn't work on a virtual pbm, cause it seems to be a > dialplan > misconfiguration. I'll let you know if I still have pbm. Thanks for > your > help. > > regards. > rod > > Tamas Cseke wrote: >> Hello, >> >> we had the same problem. we couldn't test r12581 for sure yet, but >> we will. >> This fix is only for the limit (db version), right? >> Would be limit_hash a better choice to increase performance anyway? >> Rod, do yoo have maybe experiences with limit_hash with your sipp? >> >> Thanks in advance, >> Tamas >> >> >> Mathieu Rene ?rta: >> >>> Were you doing transfers (action="transfer" in the dialplan) ? >>> >>> If yes, retry with revision 12581, and open a JIRA if it still is an >>> issue (also include a full debug log (delete your freeswitch.log >>> file >>> before doing your test and attach it after) in your JIRA) >>> >>> Math >>> >>> On 12-Mar-09, at 10:12 AM, rod wrote: >>> >>> >>> >>>> Hi list, >>>> >>>> I'm testing mod_limit like this: >>>> >>>> >>>> >>>> >>>> where AREA could be any value like: US/EUROPE/AFRICA... that has >>>> been >>>> set for the call >>>> >>>> then I use sipp for load testing with 4 cps and max 65 calls so >>>> that I >>>> will be limited by the limit of 60 in the dialplan. >>>> >>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>> value >>>> growing up to 60. >>>> >>>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>>> for >>>> each new cps in excess, FS sends a 503. >>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>> decreasing even when there is no more channels used (show >>>> channels), >>>> the >>>> limit value is stuck to 60. >>>> >>>> >>>> If I limit Sipp to 55 calls (below the limit value), the limit >>>> value >>>> increase and decrease depending on load, and the pbm doesn't >>>> appear. >>>> >>>> Does anybody is using mod_limit and have encountered the same pbm. >>>> >>>> I'm using latest svn: 12580. >>>> >>>> regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From davidwdan at gmail.com Fri Mar 13 06:45:13 2009 From: davidwdan at gmail.com (David Dan) Date: Fri, 13 Mar 2009 09:45:13 -0400 Subject: [Freeswitch-users] How do I notify FreeSwitch that a phone has been answered to play audio or TTS In-Reply-To: <8CB71BD82A0E313-CF4-158F@MBLK-M31.sysops.aol.com> References: <8CB71BD82A0E313-CF4-158F@MBLK-M31.sysops.aol.com> Message-ID: <65bd1c9f0903130645t70c12458n563319f85fe0cf3a@mail.gmail.com> Try ignore early media. On 3/13/09, mszlazak at aol.com wrote: > If I originate an outgoing call from FreeSwitch and want to tts a phrase or > play an audio once the call has been answered (i.e, someone answered their > cell phone or I got their voicemail) then how do I detect that. > > Otherwise, I've tried the following but it relies in getting the timing > right which won't always work or looping the tts phrase over and over. > > var s; > while (tryCalling()) {} > s.hangup(); > exit(); > > function tryCalling() { > ??? s = new Session("sofia/gateway/spa3102/12223334444 at 10.0.0.5:5061"); > ??? s.waitForAnswer(10000); > ??? > ??? if (s.cause == "USER_BUSY") { > ??? ?? ?return true; > ??? } > > ??? if (s.ready()) { > ??? ?? ?s.sleep(10000); > ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); > ??? } > ??? return false; > } > > > Another way is to keep replaying the tts phrase by replacing > > ? if (s.ready()) { > > ??? ?? ?s.sleep(10000); > > ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); > > ??? } > > with something like: > > while (s.ready()) { > > ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); > > ??? } > > > Thanks. Mark. > -- Sent from my mobile device From jalsot at gmail.com Fri Mar 13 06:56:24 2009 From: jalsot at gmail.com (Tamas) Date: Fri, 13 Mar 2009 14:56:24 +0100 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> Message-ID: <49BA6608.70208@gmail.com> My guess is: pbm = problem :) T. Michael Jerris ?rta: > What is a pbm? > > On Mar 13, 2009, at 4:51 AM, rod wrote: > > >> Hello, >> >> I'm now running r12590, the pbm was still there, but this was >> because of >> a broken dialplan. >> >> I'm using this for exceeded limit: >> >> > expression="^limit_exceeded$"> >> >> >> >> >> >> but this extension was at the end of my dialplan and I matched an >> other >> extension before reaching the limit extension. >> What was odd was that in ngrep I saw FS sending a 503, I will >> investigate on this. >> >> I will rerun long term test and let you know if all is ok, so I have >> to >> do for a previous mail related to ghost sessions in CLI. >> >> @Tamas: >> I just give a try to limit_hash, and didn't make many tests with it. >> It's on my todo list. Limit_hash is not a better choice than limit, >> their usage are different: >> - limit_hash is a good way to rate limit a specific gateway for >> example so that your switch won't be flooded by a misconfigured peer >> gw >> - limit is for limiting concurrent call to an extension/gateway, eg >> you have a peer that provides you with 30channels and you want to >> allow >> 15 channels for mobile (PLMN) and 15 for PSTN without relying >> >> If you still have pbm with limit and svn, pay attention to you >> dialplan :p >> >> @Mathieu >> I hope you didn't work on a virtual pbm, cause it seems to be a >> dialplan >> misconfiguration. I'll let you know if I still have pbm. Thanks for >> your >> help. >> >> regards. >> rod >> >> Tamas Cseke wrote: >> >>> Hello, >>> >>> we had the same problem. we couldn't test r12581 for sure yet, but >>> we will. >>> This fix is only for the limit (db version), right? >>> Would be limit_hash a better choice to increase performance anyway? >>> Rod, do yoo have maybe experiences with limit_hash with your sipp? >>> >>> Thanks in advance, >>> Tamas >>> >>> >>> Mathieu Rene ?rta: >>> >>> >>>> Were you doing transfers (action="transfer" in the dialplan) ? >>>> >>>> If yes, retry with revision 12581, and open a JIRA if it still is an >>>> issue (also include a full debug log (delete your freeswitch.log >>>> file >>>> before doing your test and attach it after) in your JIRA) >>>> >>>> Math >>>> >>>> On 12-Mar-09, at 10:12 AM, rod wrote: >>>> >>>> >>>> >>>> >>>>> Hi list, >>>>> >>>>> I'm testing mod_limit like this: >>>>> >>>>> >>>>> >>>>> >>>>> where AREA could be any value like: US/EUROPE/AFRICA... that has >>>>> been >>>>> set for the call >>>>> >>>>> then I use sipp for load testing with 4 cps and max 65 calls so >>>>> that I >>>>> will be limited by the limit of 60 in the dialplan. >>>>> >>>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>>> value >>>>> growing up to 60. >>>>> >>>>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>>>> for >>>>> each new cps in excess, FS sends a 503. >>>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>>> decreasing even when there is no more channels used (show >>>>> channels), >>>>> the >>>>> limit value is stuck to 60. >>>>> >>>>> >>>>> If I limit Sipp to 55 calls (below the limit value), the limit >>>>> value >>>>> increase and decrease depending on load, and the pbm doesn't >>>>> appear. >>>>> >>>>> Does anybody is using mod_limit and have encountered the same pbm. >>>>> >>>>> I'm using latest svn: 12580. >>>>> >>>>> regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Gabriel at airg.com Thu Mar 12 15:28:02 2009 From: Gabriel at airg.com (Gabriel Cho) Date: Thu, 12 Mar 2009 15:28:02 -0700 Subject: [Freeswitch-users] uuid_displace in bridged call Message-ID: <0B02E756F603CC409EB553879B090CC80A50E0C6@HPEXCHVS01.exchange.airg> Hey all, does anyone know how to detect a completion of an audio file playback using uuid_displace API in a bridged call? ie) socket.send("api uuid_bridge #{uuid} #{other_uuid}"); socket.send("api uuid_displace #{uuid} start #{filename} 0 mux"); <-- i do not receive any events from freeswitch notifying end of audio file Gabriel Cho -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/93acb4a5/attachment.html From diego.viola at gmail.com Fri Mar 13 03:46:09 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 06:46:09 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime Message-ID: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> Hello, I was wondering if it is possible to start FreeSWITCH without any SQL at all, yeah I know there is -nosql but I still want to have all the information available at run time, ie: show channels, sofia status, sofia status profile , everything. Would that improve performance? I ask this because I heard that SQLite causes some performance issues and I heard it's being removed from the core, I also think that making that optional would be nice along with UnixODBC support for everything. I also got a few situations where I had to rm -f /usr/local/freeswitch/db/* because there were some data there that was not supposed to be, that's why I think real time data on run time would be better. Anyway, just some feedback/question I wanted to share, FS is awesome, keep up the great work. Regards, Diego From diego.viola at gmail.com Fri Mar 13 03:53:01 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 06:53:01 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> Message-ID: <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> This is what I get when I do a show channels with -nosql. freeswitch at internal> show channels -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! freeswitch at internal> It would be nice if it prints the channels on runtime without the need/overheat of SQL. Regards, Diego On Fri, Mar 13, 2009 at 6:46 AM, Diego Viola wrote: > Hello, > > I was wondering if it is possible to start FreeSWITCH without any SQL > at all, yeah I know there is -nosql but I still want to have all the > information available at run time, ie: show channels, sofia status, > sofia status profile , everything. > > Would that improve performance? I ask this because I heard that SQLite > causes some performance issues and I heard it's being removed from the > core, I also think that making that optional would be nice along with > UnixODBC support for everything. > > I also got a few situations where I had to rm -f > /usr/local/freeswitch/db/* because there were some data there that was > not supposed to be, that's why I think real time data on run time > would be better. > > Anyway, just some feedback/question I wanted to share, FS is awesome, > keep up the great work. > > Regards, > > Diego > From diego.viola at gmail.com Fri Mar 13 04:12:17 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 07:12:17 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> Message-ID: <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> Erhm, this is what I wanted to say. -- I think FreeSWITCH should leave that SQLite option as optional, I think it would be nicer if FS doesn't depend on any SQL at all, if you do a show channels right now with -nosql, this is what you will see: freeswitch at internal> show channels -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! freeswitch at internal> I think it would be nice if FS can show you the information on real time, and on run time, without any SQL at all, and leave UnixODBC & SQLite as optional options, I also think this will improve performance. On Fri, Mar 13, 2009 at 6:53 AM, Diego Viola wrote: > This is what I get when I do a show channels with -nosql. > > freeswitch at internal> show channels > -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! > > freeswitch at internal> > > It would be nice if it prints the channels on runtime without the > need/overheat of SQL. > > Regards, > > Diego > > On Fri, Mar 13, 2009 at 6:46 AM, Diego Viola wrote: >> Hello, >> >> I was wondering if it is possible to start FreeSWITCH without any SQL >> at all, yeah I know there is -nosql but I still want to have all the >> information available at run time, ie: show channels, sofia status, >> sofia status profile , everything. >> >> Would that improve performance? I ask this because I heard that SQLite >> causes some performance issues and I heard it's being removed from the >> core, I also think that making that optional would be nice along with >> UnixODBC support for everything. >> >> I also got a few situations where I had to rm -f >> /usr/local/freeswitch/db/* because there were some data there that was >> not supposed to be, that's why I think real time data on run time >> would be better. >> >> Anyway, just some feedback/question I wanted to share, FS is awesome, >> keep up the great work. >> >> Regards, >> >> Diego >> > From diego.viola at gmail.com Fri Mar 13 05:56:46 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 08:56:46 -0400 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! In-Reply-To: <8CB716FC809DF3B-8FC-1005@webmail-mh34.sysops.aol.com> References: <8CB716FC809DF3B-8FC-1005@webmail-mh34.sysops.aol.com> Message-ID: <86a32abc0903130556r5ea96a6anb47eb99b7052756b@mail.gmail.com> Good initiative, congrats. I already registered to it. Diego On Thu, Mar 12, 2009 at 5:22 PM, wrote: > Great, I like forums better than lists. Make sure the folks at FreeSwitch > make it know to anyone coming to any of their pages by providing links, etc. > Thanks > > > -----Original Message----- > From: Harry FSwitch > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, 12 Mar 2009 11:55 am > Subject: [Freeswitch-users] FreeSWITCH forum community opened today! > > Greetings, > > Last week I submitted this post to the mailing list... > > http://www.nabble.com/Please-end-the-torment-td22352222.html > > I received a mixed response to say the least, no one however emailed me > saying they wanted to be involved or would contribute hosting resources. I > figured eh screw-it, and went ahead and created the forums anyway and opened > them today! > > http://freeswitch411.info > > I went on IRC to let folks know its available and after a few jabs from the > crowd Anthony said: > "#freeswitch 2009-03-12 10:47:04 [anthm] harr, if you maintain it you are > welcome to have it" > > If it takes off and provides a friendly, helpful entryway for new FreeSWITCH > users then I will be happy, if it flops I'll be said. But I will maintain it > and do what I can to help FreeSWITCH grow. This is the only time I'll > overtly mention it on this list, I'll have the link in my signature and > thats about it. :) > > Thanks for your attention > > -- > Harry > http://freeswitch411.info > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ________________________________ > A Good Credit Score is 700 or Above. See yours in just 2 easy steps! > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From rupa at rupa.com Fri Mar 13 07:05:33 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 13 Mar 2009 09:05:33 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> Message-ID: Using sql for the show commands is pretty elegant. It allows one to get info without having to lock a bunch of in memory data structures prior to collecting stats. Locking data structures just for reporting == bad -- especially on a high volume server. On Fri, Mar 13, 2009 at 6:12 AM, Diego Viola wrote: > Erhm, this is what I wanted to say. > > -- > > I think FreeSWITCH should leave that SQLite option as optional, I > think it would be nicer if FS doesn't depend on any SQL at all, if you > do a show channels right now with -nosql, this is what you will see: > > freeswitch at internal> show channels > -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! > > freeswitch at internal> > > I think it would be nice if FS can show you the information on real > time, and on run time, without any SQL at all, and leave UnixODBC & > SQLite as optional options, I also think this will improve > performance. > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/bf0a7d26/attachment.html From mrene_lists at avgs.ca Fri Mar 13 07:09:05 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 13 Mar 2009 10:09:05 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> Message-ID: <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> In this case sqlite separates the core from what show channels uses etc, the DB is filled in by an event handler. This actually increases performance since you dont need to exclusively lock the session manager's table just to list channels. Math On 13-Mar-09, at 7:12 AM, Diego Viola wrote: > Erhm, this is what I wanted to say. > > -- > > I think FreeSWITCH should leave that SQLite option as optional, I > think it would be nicer if FS doesn't depend on any SQL at all, if you > do a show channels right now with -nosql, this is what you will see: > > freeswitch at internal> show channels > -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! > > freeswitch at internal> > > I think it would be nice if FS can show you the information on real > time, and on run time, without any SQL at all, and leave UnixODBC & > SQLite as optional options, I also think this will improve > performance. > > On Fri, Mar 13, 2009 at 6:53 AM, Diego Viola > wrote: >> This is what I get when I do a show channels with -nosql. >> >> freeswitch at internal> show channels >> -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! >> >> freeswitch at internal> >> >> It would be nice if it prints the channels on runtime without the >> need/overheat of SQL. >> >> Regards, >> >> Diego >> >> On Fri, Mar 13, 2009 at 6:46 AM, Diego Viola >> wrote: >>> Hello, >>> >>> I was wondering if it is possible to start FreeSWITCH without any >>> SQL >>> at all, yeah I know there is -nosql but I still want to have all the >>> information available at run time, ie: show channels, sofia status, >>> sofia status profile , everything. >>> >>> Would that improve performance? I ask this because I heard that >>> SQLite >>> causes some performance issues and I heard it's being removed from >>> the >>> core, I also think that making that optional would be nice along >>> with >>> UnixODBC support for everything. >>> >>> I also got a few situations where I had to rm -f >>> /usr/local/freeswitch/db/* because there were some data there that >>> was >>> not supposed to be, that's why I think real time data on run time >>> would be better. >>> >>> Anyway, just some feedback/question I wanted to share, FS is >>> awesome, >>> keep up the great work. >>> >>> Regards, >>> >>> Diego >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Mark.Tabron at rnid-typetalk.org.uk Fri Mar 13 07:16:14 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Fri, 13 Mar 2009 14:16:14 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> Message-ID: <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! http://pastebin.freeswitch.org/7751 Will setup an IRC client and see if I can log onto the channel. Thanks again! -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 March 2009 16:50 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 > My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so > apologies if some of my terminology isn't quite correct. Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > > > > Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've > hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the > card is green and wanrouter is installed and up in TDM_API mode, with the > connection status showing as connected. ?Configured Openzap for 9 b and 1 d > channel as described in Freeswitch Wiki. Then created a diaplan to fire off > any calls preceded by 9 to the next available openzap channel. Looks good so far... > The problem I have is when I initiate an external call (using 9xxxxxxx) from > an extension I can see Freeswitch allocating the call to the next available > channel but then the just sits there and times out after 1 minute. With the > cause stated as ORIGINATOR_CANCEL (guessing this is the time out) okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has lots of useful information for how to gather log information, how to use the pastebin, etc. Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers: To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org). Welcome to the wonderful world of telephony debugging! -MC P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.) _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- From intralanman at freeswitch.org Fri Mar 13 09:05:32 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 13 Mar 2009 12:05:32 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA6608.70208@gmail.com> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> <49BA6608.70208@gmail.com> Message-ID: <49BA844C.3010409@freeswitch.org> Tamas wrote: > My guess is: pbm = problem :) > sure, but is it really that hard to spell all the way out? -Ray From intralanman at freeswitch.org Fri Mar 13 09:09:39 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 13 Mar 2009 12:09:39 -0400 Subject: [Freeswitch-users] Load testing on XP In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380499DD68@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF461380499DD68@nickel.mettonigroup.com> Message-ID: <49BA8543.5070401@freeswitch.org> Richard Lamkin wrote: > > Dear All, > > > > Does anyone have any load test results using XP(SP3) -- I have checked > the wiki but could not find any recorded. > > > > Regards > > > > Richard Lamkin > for the most part, we try to steer clear of posting load test results since my testing may be different than yours, it's hard (if not impossible) for two separate users to get the same results. so we recommend you try your own load testing the mirrors your scenario specifically. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/8d11e376/attachment.html From cstomi.levlist at gmail.com Fri Mar 13 09:19:00 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Fri, 13 Mar 2009 17:19:00 +0100 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA1EA5.4050201@laposte.net> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> Message-ID: <49BA8774.3040803@gmail.com> Hello, Thank you for your help. limit,Limit, [ [number [dialplan [context]]]] limit_hash,Limit (hash), [[/interval]] [number [dialplan [context]]] I diged into mod_limit deeplier and as far as I understand, limit and limit_hash can be used for the same thing: limiting concurrent calls. please correct me, if I'm wrong... but limit_hash has another feature /interval this is that make the rate limitation you mentioned. Our dialplan is good that is out of question. I see hangup in console. we do I tested both of them with sipp and I wasn't able to reproduce the problem r12581 did fix the case when limit is called more times, right? Thanks, Tamas rod ?rta: > Hello, > > I'm now running r12590, the pbm was still there, but this was because of > a broken dialplan. > > I'm using this for exceeded limit: > > > > > > > > but this extension was at the end of my dialplan and I matched an other > extension before reaching the limit extension. > What was odd was that in ngrep I saw FS sending a 503, I will > investigate on this. > > I will rerun long term test and let you know if all is ok, so I have to > do for a previous mail related to ghost sessions in CLI. > > @Tamas: > I just give a try to limit_hash, and didn't make many tests with it. > It's on my todo list. Limit_hash is not a better choice than limit, > their usage are different: > - limit_hash is a good way to rate limit a specific gateway for > example so that your switch won't be flooded by a misconfigured peer gw > - limit is for limiting concurrent call to an extension/gateway, eg > you have a peer that provides you with 30channels and you want to allow > 15 channels for mobile (PLMN) and 15 for PSTN without relying > > If you still have pbm with limit and svn, pay attention to you dialplan :p > > @Mathieu > I hope you didn't work on a virtual pbm, cause it seems to be a dialplan > misconfiguration. I'll let you know if I still have pbm. Thanks for your > help. > > regards. > rod > > Tamas Cseke wrote: > >> Hello, >> >> we had the same problem. we couldn't test r12581 for sure yet, but we will. >> This fix is only for the limit (db version), right? >> Would be limit_hash a better choice to increase performance anyway? >> Rod, do yoo have maybe experiences with limit_hash with your sipp? >> >> Thanks in advance, >> Tamas >> >> >> Mathieu Rene ?rta: >> >> >>> Were you doing transfers (action="transfer" in the dialplan) ? >>> >>> If yes, retry with revision 12581, and open a JIRA if it still is an >>> issue (also include a full debug log (delete your freeswitch.log file >>> before doing your test and attach it after) in your JIRA) >>> >>> Math >>> >>> On 12-Mar-09, at 10:12 AM, rod wrote: >>> >>> >>> >>> >>>> Hi list, >>>> >>>> I'm testing mod_limit like this: >>>> >>>> >>>> >>>> >>>> where AREA could be any value like: US/EUROPE/AFRICA... that has been >>>> set for the call >>>> >>>> then I use sipp for load testing with 4 cps and max 65 calls so that I >>>> will be limited by the limit of 60 in the dialplan. >>>> >>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>> value >>>> growing up to 60. >>>> >>>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>>> for >>>> each new cps in excess, FS sends a 503. >>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>> decreasing even when there is no more channels used (show channels), >>>> the >>>> limit value is stuck to 60. >>>> >>>> >>>> If I limit Sipp to 55 calls (below the limit value), the limit value >>>> increase and decrease depending on load, and the pbm doesn't appear. >>>> >>>> Does anybody is using mod_limit and have encountered the same pbm. >>>> >>>> I'm using latest svn: 12580. >>>> >>>> regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mrene_lists at avgs.ca Fri Mar 13 09:26:05 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 13 Mar 2009 12:26:05 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA8774.3040803@gmail.com> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <49BA8774.3040803@gmail.com> Message-ID: Yeah, you should be using limit_hash unless you need data replication on multiple FS boxes since its a lot faster. You can also do On another note, normal_circuit_congestion looks like a more appropriate cause. If you have problems, please open a jira including logs and a detailed description of the required steps to reproduce the problem. Math On 13-Mar-09, at 12:19 PM, Tamas Cseke wrote: > Hello, > > Thank you for your help. > > limit,Limit, [ [number [dialplan [context]]]] > limit_hash,Limit (hash), [[/interval]] [number > [dialplan [context]]] > > I diged into mod_limit deeplier and as far as I understand, limit and > limit_hash can be used for the same thing: > limiting concurrent calls. please correct me, if I'm wrong... > > but limit_hash has another feature /interval this is that make the > rate > limitation you mentioned. > > Our dialplan is good that is out of question. I see hangup in console. > we do > > I tested both of them with sipp and I wasn't able to reproduce the > problem > > r12581 did fix the case when limit is called more times, right? > > Thanks, > Tamas > > > rod ?rta: >> Hello, >> >> I'm now running r12590, the pbm was still there, but this was >> because of >> a broken dialplan. >> >> I'm using this for exceeded limit: >> >> > expression="^limit_exceeded$"> >> >> >> >> >> >> but this extension was at the end of my dialplan and I matched an >> other >> extension before reaching the limit extension. >> What was odd was that in ngrep I saw FS sending a 503, I will >> investigate on this. >> >> I will rerun long term test and let you know if all is ok, so I >> have to >> do for a previous mail related to ghost sessions in CLI. >> >> @Tamas: >> I just give a try to limit_hash, and didn't make many tests with it. >> It's on my todo list. Limit_hash is not a better choice than limit, >> their usage are different: >> - limit_hash is a good way to rate limit a specific gateway for >> example so that your switch won't be flooded by a misconfigured >> peer gw >> - limit is for limiting concurrent call to an extension/gateway, >> eg >> you have a peer that provides you with 30channels and you want to >> allow >> 15 channels for mobile (PLMN) and 15 for PSTN without relying >> >> If you still have pbm with limit and svn, pay attention to you >> dialplan :p >> >> @Mathieu >> I hope you didn't work on a virtual pbm, cause it seems to be a >> dialplan >> misconfiguration. I'll let you know if I still have pbm. Thanks for >> your >> help. >> >> regards. >> rod >> >> Tamas Cseke wrote: >> >>> Hello, >>> >>> we had the same problem. we couldn't test r12581 for sure yet, but >>> we will. >>> This fix is only for the limit (db version), right? >>> Would be limit_hash a better choice to increase performance anyway? >>> Rod, do yoo have maybe experiences with limit_hash with your sipp? >>> >>> Thanks in advance, >>> Tamas >>> >>> >>> Mathieu Rene ?rta: >>> >>> >>>> Were you doing transfers (action="transfer" in the dialplan) ? >>>> >>>> If yes, retry with revision 12581, and open a JIRA if it still is >>>> an >>>> issue (also include a full debug log (delete your freeswitch.log >>>> file >>>> before doing your test and attach it after) in your JIRA) >>>> >>>> Math >>>> >>>> On 12-Mar-09, at 10:12 AM, rod wrote: >>>> >>>> >>>> >>>> >>>>> Hi list, >>>>> >>>>> I'm testing mod_limit like this: >>>>> >>>>> >>>>> >>>>> >>>>> where AREA could be any value like: US/EUROPE/AFRICA... that has >>>>> been >>>>> set for the call >>>>> >>>>> then I use sipp for load testing with 4 cps and max 65 calls so >>>>> that I >>>>> will be limited by the limit of 60 in the dialplan. >>>>> >>>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>>> value >>>>> growing up to 60. >>>>> >>>>> Sipp still tries to establish new call (up to 65 calls) at 4cps >>>>> and >>>>> for >>>>> each new cps in excess, FS sends a 503. >>>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>>> decreasing even when there is no more channels used (show >>>>> channels), >>>>> the >>>>> limit value is stuck to 60. >>>>> >>>>> >>>>> If I limit Sipp to 55 calls (below the limit value), the limit >>>>> value >>>>> increase and decrease depending on load, and the pbm doesn't >>>>> appear. >>>>> >>>>> Does anybody is using mod_limit and have encountered the same pbm. >>>>> >>>>> I'm using latest svn: 12580. >>>>> >>>>> regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cstomi.levlist at gmail.com Fri Mar 13 09:17:46 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Fri, 13 Mar 2009 17:17:46 +0100 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA1EA5.4050201@laposte.net> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> Message-ID: <49BA872A.9030003@gmail.com> Hello, Thank you for your help. limit,Limit, [ [number [dialplan [context]]]] limit_hash,Limit (hash), [[/interval]] [number [dialplan [context]]] I diged into mod_limit deeplier and as far as I understand, limit and limit_hash can be used for the same thing: limiting concurrent calls. please correct me, if I'm wrong... but limit_hash has another feature /interval this is that make the rate limitation you mentioned. Our dialplan is good that is out of question. I see hangup in console. we do I tested both of them with sipp and I wasn't able to reproduce the problem r12581 did fix the case when limit is called more times, right? Thanks, Tamas rod ?rta: > Hello, > > I'm now running r12590, the pbm was still there, but this was because of > a broken dialplan. > > I'm using this for exceeded limit: > > > > > > > > but this extension was at the end of my dialplan and I matched an other > extension before reaching the limit extension. > What was odd was that in ngrep I saw FS sending a 503, I will > investigate on this. > > I will rerun long term test and let you know if all is ok, so I have to > do for a previous mail related to ghost sessions in CLI. > > @Tamas: > I just give a try to limit_hash, and didn't make many tests with it. > It's on my todo list. Limit_hash is not a better choice than limit, > their usage are different: > - limit_hash is a good way to rate limit a specific gateway for > example so that your switch won't be flooded by a misconfigured peer gw > - limit is for limiting concurrent call to an extension/gateway, eg > you have a peer that provides you with 30channels and you want to allow > 15 channels for mobile (PLMN) and 15 for PSTN without relying > > If you still have pbm with limit and svn, pay attention to you dialplan :p > > @Mathieu > I hope you didn't work on a virtual pbm, cause it seems to be a dialplan > misconfiguration. I'll let you know if I still have pbm. Thanks for your > help. > > regards. > rod > > Tamas Cseke wrote: > >> Hello, >> >> we had the same problem. we couldn't test r12581 for sure yet, but we will. >> This fix is only for the limit (db version), right? >> Would be limit_hash a better choice to increase performance anyway? >> Rod, do yoo have maybe experiences with limit_hash with your sipp? >> >> Thanks in advance, >> Tamas >> >> >> Mathieu Rene ?rta: >> >> >>> Were you doing transfers (action="transfer" in the dialplan) ? >>> >>> If yes, retry with revision 12581, and open a JIRA if it still is an >>> issue (also include a full debug log (delete your freeswitch.log file >>> before doing your test and attach it after) in your JIRA) >>> >>> Math >>> >>> On 12-Mar-09, at 10:12 AM, rod wrote: >>> >>> >>> >>> >>>> Hi list, >>>> >>>> I'm testing mod_limit like this: >>>> >>>> >>>> >>>> >>>> where AREA could be any value like: US/EUROPE/AFRICA... that has been >>>> set for the call >>>> >>>> then I use sipp for load testing with 4 cps and max 65 calls so that I >>>> will be limited by the limit of 60 in the dialplan. >>>> >>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>> value >>>> growing up to 60. >>>> >>>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>>> for >>>> each new cps in excess, FS sends a 503. >>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>> decreasing even when there is no more channels used (show channels), >>>> the >>>> limit value is stuck to 60. >>>> >>>> >>>> If I limit Sipp to 55 calls (below the limit value), the limit value >>>> increase and decrease depending on load, and the pbm doesn't appear. >>>> >>>> Does anybody is using mod_limit and have encountered the same pbm. >>>> >>>> I'm using latest svn: 12580. >>>> >>>> regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Fri Mar 13 07:36:20 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 10:36:20 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> Message-ID: <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> Oh, I thought that SQLite wasn't that great on performance and that people wanted to replace/remove it from the core. "On of the most interesting things about FreeSWITCH to me has been the fact that most data in the system such as registrations are kept in a SQL database. The default installation uses SQLite internally though you can easily point FreeSWITCH at one of a number of other SQL servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has become somewhat of a bottleneck in the core so future versions of FreeSWITCH will use less of it. For example, doing a "show channels" with over 500 channels in use starts to show issues. While I'm sad to see SQLite go in these cases, I am anxious to see how Minessale replaces it." http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale I was just being curious about it :-) Regards, Diego On Fri, Mar 13, 2009 at 10:09 AM, Mathieu Rene wrote: > In this case sqlite separates the core from what show channels uses > etc, the DB is filled in by an event handler. This actually increases > performance since you dont need to exclusively lock the session > manager's table just to list channels. > > > Math > > On 13-Mar-09, at 7:12 AM, Diego Viola wrote: > >> Erhm, this is what I wanted to say. >> >> -- >> >> I think FreeSWITCH should leave that SQLite option as optional, I >> think it would be nicer if FS doesn't depend on any SQL at all, if you >> do a show channels right now with -nosql, this is what you will see: >> >> freeswitch at internal> show channels >> -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! >> >> freeswitch at internal> >> >> I think it would be nice if FS can show you the information on real >> time, and on run time, without any SQL at all, and leave UnixODBC & >> SQLite as optional options, I also think this will improve >> performance. >> >> On Fri, Mar 13, 2009 at 6:53 AM, Diego Viola >> wrote: >>> This is what I get when I do a show channels with -nosql. >>> >>> freeswitch at internal> show channels >>> -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! >>> >>> freeswitch at internal> >>> >>> It would be nice if it prints the channels on runtime without the >>> need/overheat of SQL. >>> >>> Regards, >>> >>> Diego >>> >>> On Fri, Mar 13, 2009 at 6:46 AM, Diego Viola >>> wrote: >>>> Hello, >>>> >>>> I was wondering if it is possible to start FreeSWITCH without any >>>> SQL >>>> at all, yeah I know there is -nosql but I still want to have all the >>>> information available at run time, ie: show channels, sofia status, >>>> sofia status profile , everything. >>>> >>>> Would that improve performance? I ask this because I heard that >>>> SQLite >>>> causes some performance issues and I heard it's being removed from >>>> the >>>> core, I also think that making that optional would be nice along >>>> with >>>> UnixODBC support for everything. >>>> >>>> I also got a few situations where I had to rm -f >>>> /usr/local/freeswitch/db/* because there were some data there that >>>> was >>>> not supposed to be, that's why I think real time data on run time >>>> would be better. >>>> >>>> Anyway, just some feedback/question I wanted to share, FS is >>>> awesome, >>>> keep up the great work. >>>> >>>> Regards, >>>> >>>> Diego >>>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Mar 13 12:02:42 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Mar 2009 14:02:42 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> Message-ID: <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> Since we added indexes to the SQLite DB its not so bad. /b On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > Oh, I thought that SQLite wasn't that great on performance and that > people wanted to replace/remove it from the core. > > "On of the most interesting things about FreeSWITCH to me has been the > fact that most data in the system such as registrations are kept in a > SQL database. The default installation uses SQLite internally though > you can easily point FreeSWITCH at one of a number of other SQL > servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has > become somewhat of a bottleneck in the core so future versions of > FreeSWITCH will use less of it. For example, doing a "show channels" > with over 500 channels in use starts to show issues. While I'm sad to > see SQLite go in these cases, I am anxious to see how Minessale > replaces it." > > http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > > I was just being curious about it :-) > > Regards, > > Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/79793daf/attachment.html From Prometheus001 at gmx.net Fri Mar 13 12:13:47 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 13 Mar 2009 20:13:47 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: References: Message-ID: <49BAB06B.2060009@gmx.net> Ok, I tried it. It works fine on dialthru. SDP header is enhanced. Perfect! However I have a problem which I already discussed before in this round, and I am very close to solve it, I think. Scenario is as follows * A callback application which is done through event socket calls party A with cid-number 'unknown' * Party A enters Digits which represents the number of party B * I set the caller-id-no and effective-caller-id-no to the value of A and sip_cid_type=rpid * Call is then transferred from A to Party B via the xml dialplan * Now I have 2 different variables when I trace the event_socket o caller-caller-id-number is 'anonymous' and this is submitted in the INVITE message to B o variable-caller-id-number is the right number of B, this is not part of the Sip message * B sees 'unknown' as the number of party A. This is also represented in the SIP mssage sent to B * I want B to see the right number of party A My question: What can I do to set to in the XML dialplan. Here is the part of the dialplan where I want so set it: Anybody has a solution? Best regards Peter Ken Rice schrieb: > These should be available any time you are going to process a call thru the > dialplan and call a bridge on the call > > > >> From: Peter P GMX >> Reply-To: >> Date: Tue, 10 Mar 2009 23:20:39 +0100 >> To: >> Subject: Re: [Freeswitch-users] Rewriting Remote Party ID >> >> Hello, >> >> are these variables only available at call setup time or can they be >> changed during a call, e.g. before a call is being transferred to >> another destination? >> >> Best regards >> Peter >> >> Michael Collins schrieb: >> >>> On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale >>> wrote: >>> >>> >>>> Latest SVN: >>>> >>>> Send no extra caller id info: >>>> {sip_cid_type=none}sofia/default/user at example.com >>>> >>>> Send RPID (default) >>>> {sip_cid_type=rpid}sofia/default/user at example.com >>>> >>>> Send P-XXX-Identity >>>> {sip_cid_type=pid}sofia/default/user at example.com >>>> >>>> Send RPID with chosen content >>>> {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_num >>>> ber=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul >>>> tuser at example.com >>>> >>>> >>> FYI, I added this info to the channel variables page: >>> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type >>> >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Fri Mar 13 12:57:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Mar 2009 12:57:17 -0700 Subject: [Freeswitch-users] New Voice Prompts - Need Community's Help Message-ID: <87f2f3b90903131257x78e763fdw55015566711cbb60@mail.gmail.com> FYI, I've just started a new JIRA for the new voice prompts list that we are getting together: http://jira.freeswitch.org/browse/FSSCRIPTS-15 Please add comments to this JIRA if you can think of useful prompts for FreeSWITCH that we don't already have recorded. We also need financial support to get these recorded. All those who wish to assist, please donate to brian at freeswitch.org on Paypal. Thanks again to all of you who make FreeSWITCH such a great community! -MC From gkuri at ieee.org Fri Mar 13 13:36:58 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 13 Mar 2009 13:36:58 -0700 Subject: [Freeswitch-users] inband dtmf detection Message-ID: <49BAC3EA.5030404@ieee.org> shoot me if I'm on the wrong track, but is it possible to use the start_dtmf app to do inband dtmf detection and "convert" the inband DTMF to rfc2833, as opposed to using the dtmf detection on a Linksys or Grandstream ATA? the reason I ask is the dtmf detection on these ATAs seems to falsely catch dtmf tones during certain conversations and falsely inject a dtmf tone into the stream (I believe the appropriate term is "talk off"). So I was thinking of setting the ATAs to do inband DTMF rather than the current rfc2833 setting, and having FS do the more reliable dtmf detection instead? Thanks, Gabe From brian at freeswitch.org Fri Mar 13 13:44:16 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Mar 2009 15:44:16 -0500 Subject: [Freeswitch-users] inband dtmf detection In-Reply-To: <49BAC3EA.5030404@ieee.org> References: <49BAC3EA.5030404@ieee.org> Message-ID: <58A70382-5FE1-4C26-B8B3-F161967F39E6@freeswitch.org> That is the purpose of start_dtmf to detect it inband and convert it to 2833. Unless you disabled 2833. /b On Mar 13, 2009, at 3:36 PM, Gabriel Kuri wrote: > shoot me if I'm on the wrong track, but is it possible to use the > start_dtmf app to do inband dtmf detection and "convert" the inband > DTMF > to rfc2833, as opposed to using the dtmf detection on a Linksys or > Grandstream ATA? From gkuri at ieee.org Fri Mar 13 14:33:45 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 13 Mar 2009 14:33:45 -0700 Subject: [Freeswitch-users] inband dtmf detection In-Reply-To: <58A70382-5FE1-4C26-B8B3-F161967F39E6@freeswitch.org> References: <49BAC3EA.5030404@ieee.org> <58A70382-5FE1-4C26-B8B3-F161967F39E6@freeswitch.org> Message-ID: <49BAD139.50907@ieee.org> thanks, I'll give it a try. I'm assuming this app requires a reasonable amount of additional processing power, particularly on a box with several hundred active channels, as it's basically snooping the rtp stream looking for inband dtmf tones? Gabe Brian West wrote: > That is the purpose of start_dtmf to detect it inband and convert it > to 2833. Unless you disabled 2833. > > /b > > On Mar 13, 2009, at 3:36 PM, Gabriel Kuri wrote: > >> shoot me if I'm on the wrong track, but is it possible to use the >> start_dtmf app to do inband dtmf detection and "convert" the inband >> DTMF >> to rfc2833, as opposed to using the dtmf detection on a Linksys or >> Grandstream ATA? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mszlazak at aol.com Fri Mar 13 16:38:28 2009 From: mszlazak at aol.com (mszlazak) Date: Fri, 13 Mar 2009 16:38:28 -0700 (PDT) Subject: [Freeswitch-users] Help with detecting hangup events. Message-ID: <22507114.post@talk.nabble.com> I have a problem in setting up an extension that will detect different hangup events. If an outside call comes in and gets sent to "myExtension" then that call is bridged to an application. When the application hangs up then the hangup hook "notify.js" is used to originate a call to another phone which notifies that phone that the application has been used. However, the problem here is that in the initial call, the caller can't hangup because then notify.js won't execute. I haven't tried this but if I changed things so "hang_after_bridge=false" then I'm guessing that notify.js will execute when the initial caller hangs up their phone. This I want but I don't want the call originated in notify.js to happen unless the bridged to application also issued a hangup previously. Otherwise, if the caller hangs up before the application does it's thing and hasn't issued a hangup then notify.js will still originate calls and these I don't want. So, I want notify.js to originate calls after the caller hangs up but these hangups have to be those that came after the bridged to application issued a hung up. I hope this makes sense. Thanks. Mark. -- View this message in context: http://www.nabble.com/Help-with-detecting-hangup-events.-tp22507114p22507114.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From diego.viola at gmail.com Fri Mar 13 21:12:28 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 14 Mar 2009 00:12:28 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> Message-ID: <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> Yeah, but still, it would be nice to see the channels with -nosql :) I don't want to be a pain in the ass, just giving some user feedback. Regards, Diego On Fri, Mar 13, 2009 at 3:02 PM, Brian West wrote: > Since we added indexes to the SQLite DB its not so bad. > /b > On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > > Oh, I thought that SQLite wasn't that great on performance and that > people wanted to replace/remove it from the core. > > "On of the most interesting things about FreeSWITCH to me has been the > fact that most data in the system such as registrations are kept in a > SQL database. The default installation uses SQLite internally though > you can easily point FreeSWITCH at one of a number of other SQL > servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has > become somewhat of a bottleneck in the core so future versions of > FreeSWITCH will use less of it. For example, doing a "show channels" > with over 500 channels in use starts to show issues. While I'm sad to > see SQLite go in these cases, I am anxious to see how Minessale > replaces it." > > http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > > I was just being curious about it :-) > > Regards, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From yudha2008 at gmail.com Sat Mar 14 00:42:48 2009 From: yudha2008 at gmail.com (Baskar) Date: Sat, 14 Mar 2009 13:12:48 +0530 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: <49BA1D6F.30002@gmail.com> References: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> <49BA1D6F.30002@gmail.com> Message-ID: *Hi, I am using TDMAPI with EuroISDN. * *My openzap.conf* *[span wanpipe] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 [span wanpipe] name => OpenZAP number => 2 trunk_type => E1 b-channel => 2:1-15 d-channel => 2:16 b-channel => 2:17-31 * *My openzap.con.xml* * ** ** * * -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090314/0bbbb77d/attachment.html From jp.manchu at gmail.com Sat Mar 14 04:49:02 2009 From: jp.manchu at gmail.com (JayaPrakash) Date: Sat, 14 Mar 2009 17:19:02 +0530 Subject: [Freeswitch-users] Issue relating mod_nibblebill Message-ID: Hi All, I am newbie to Freeswitch. I installed freeswitch-1.0.3 in Debian machine. I am able to make call, check presence, retrieve CDRs. I followed the installation steps given in mod_nibblebill for rating. While, installing mysql-connector-odbc, it has thrown errors related to mysql-config file, that it does not exist. Coming to mysql, mysql-client-5 and mysql-server-5 are installed. So I installed libmyodbc which is used for the same functionality. Rest of the steps are done, as given in mod-nibblebill installation. When the freeradius server is restarted, it has given the following error. 2009-03-14 14:55:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-03-14 14:55:47 [CRIT] mod_nibblebill.c:233 load_config() Cannot connect to ODBC driver/database freeswitch (user: root / pass dev)! Will you please have a look in solving this issue ? , how the issue can be solved? Thanks & Regards, JP. From mike at jerris.com Sat Mar 14 06:20:51 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Mar 2009 09:20:51 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> Message-ID: To clarify, -nosql turns on and off only the collecting of data for the show commands, and can now handle higher load than the sip stack can. The only thing your doing by saying -nosql is turning off the exact functionality you say you want. Its similar to saying I would like to support sip but don't want to load mod_sofia. There should be no reasons to use that command anymore, if you encounter any I would be interested in knowing what is going on. Mike On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > Yeah, but still, it would be nice to see the channels with -nosql :) > > I don't want to be a pain in the ass, just giving some user feedback. > > Regards, > > Diego > > On Fri, Mar 13, 2009 at 3:02 PM, Brian West > wrote: >> Since we added indexes to the SQLite DB its not so bad. >> /b >> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >> >> Oh, I thought that SQLite wasn't that great on performance and that >> people wanted to replace/remove it from the core. >> >> "On of the most interesting things about FreeSWITCH to me has been >> the >> fact that most data in the system such as registrations are kept in a >> SQL database. The default installation uses SQLite internally though >> you can easily point FreeSWITCH at one of a number of other SQL >> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >> become somewhat of a bottleneck in the core so future versions of >> FreeSWITCH will use less of it. For example, doing a "show channels" >> with over 500 channels in use starts to show issues. While I'm sad to >> see SQLite go in these cases, I am anxious to see how Minessale >> replaces it." >> >> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >> >> I was just being curious about it :-) >> >> Regards, >> >> Diego >> From Prometheus001 at gmx.net Sat Mar 14 06:29:40 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 14 Mar 2009 14:29:40 +0100 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: References: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> <49BA1D6F.30002@gmail.com> Message-ID: <49BBB144.60300@gmx.net> Hello Baskar, I had the same once. Please use in your configs. Euro dialect did not work any more for me in newer versions of freeswitch. Best regards Peter Baskar schrieb: > *Hi, > > I am using TDMAPI with EuroISDN. > * > *My openzap.conf* > > *[span wanpipe] > name => OpenZAP > number => 1 > trunk_type => E1 > b-channel => 1:1-15 > d-channel => 1:16 > b-channel => 1:17-31 > > [span wanpipe] > name => OpenZAP > number => 2 > trunk_type => E1 > b-channel => 2:1-15 > d-channel => 2:16 > b-channel => 2:17-31 > * > > *My openzap.con.xml* > * > > > > > > > > > > > > ** > > > > > ** > > * > * > > -- > Warm Regards, > N.Baskar > * > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Sat Mar 14 06:31:15 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 14 Mar 2009 14:31:15 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49BAB06B.2060009@gmx.net> References: <49BAB06B.2060009@gmx.net> Message-ID: <49BBB1A3.5010708@gmx.net> I got it to work. It was just so easy: Best regards Peter Peter P GMX schrieb: > Ok, I tried it. It works fine on dialthru. SDP header is enhanced. Perfect! > > However I have a problem which I already discussed before in this round, > and I am very close to solve it, I think. > Scenario is as follows > > * A callback application which is done through event socket calls > party A with cid-number 'unknown' > * Party A enters Digits which represents the number of party B > * I set the caller-id-no and effective-caller-id-no to the value of > A and sip_cid_type=rpid > * Call is then transferred from A to Party B via the xml dialplan > * Now I have 2 different variables when I trace the event_socket > o caller-caller-id-number is 'anonymous' and this is submitted > in the INVITE message to B > o variable-caller-id-number is the right number of B, this is > not part of the Sip message > * B sees 'unknown' as the number of party A. This is also > represented in the SIP mssage sent to B > * I want B to see the right number of party A > > My question: What can I do to set to > in the XML dialplan. > > Here is the part of the dialplan where I want so set it: > > > > > > > > > > Anybody has a solution? > > Best regards > Peter > > > > > Ken Rice schrieb: > >> These should be available any time you are going to process a call thru the >> dialplan and call a bridge on the call >> >> >> >> >>> From: Peter P GMX >>> Reply-To: >>> Date: Tue, 10 Mar 2009 23:20:39 +0100 >>> To: >>> Subject: Re: [Freeswitch-users] Rewriting Remote Party ID >>> >>> Hello, >>> >>> are these variables only available at call setup time or can they be >>> changed during a call, e.g. before a call is being transferred to >>> another destination? >>> >>> Best regards >>> Peter >>> >>> Michael Collins schrieb: >>> >>> >>>> On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale >>>> wrote: >>>> >>>> >>>> >>>>> Latest SVN: >>>>> >>>>> Send no extra caller id info: >>>>> {sip_cid_type=none}sofia/default/user at example.com >>>>> >>>>> Send RPID (default) >>>>> {sip_cid_type=rpid}sofia/default/user at example.com >>>>> >>>>> Send P-XXX-Identity >>>>> {sip_cid_type=pid}sofia/default/user at example.com >>>>> >>>>> Send RPID with chosen content >>>>> {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_num >>>>> ber=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul >>>>> tuser at example.com >>>>> >>>>> >>>>> >>>> FYI, I added this info to the channel variables page: >>>> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type >>>> >>>> -MC >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From rupa at rupa.com Sat Mar 14 07:08:24 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Sat, 14 Mar 2009 09:08:24 -0500 Subject: [Freeswitch-users] Issue relating mod_nibblebill In-Reply-To: References: Message-ID: Have you configured unixodbc? http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc covers configuring odbc in detail. On Sat, Mar 14, 2009 at 6:49 AM, JayaPrakash wrote: > 2009-03-14 14:55:47 [ERR] switch_odbc.c:164 > switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: > [unixODBC][Driver Manager]Data source name not found, and no default > driver specified > This means unixODBC couldn't find the datasource you specified in it's configuration file. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090314/50d6b70d/attachment.html From anthony.minessale at gmail.com Sat Mar 14 07:15:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Mar 2009 09:15:46 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B8E0AC.6070307@laposte.net> References: <49AFC1C3.9030603@laposte.net> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> <49B8E0AC.6070307@laposte.net> Message-ID: <191c3a030903140715g3a77571bi45268ee36d632771@mail.gmail.com> origination_privacy was indeed broken fixed in r12603 On Thu, Mar 12, 2009 at 5:15 AM, rod wrote: > Thanks a lot Anthony, > > it's working great. > I'm just checking the origination_privacy parameter, cause it seems to > do nothing in my setup. > > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090314/a0ae9dac/attachment.html From tayeb.meftah at gmail.com Sat Mar 14 09:47:36 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sat, 14 Mar 2009 17:47:36 +0100 Subject: [Freeswitch-users] Getting a free Did number for my FS Message-ID: <49BBDFA8.5010805@gmail.com> hello, please ho to get a free did number ? also, is it pocible to link it to my FS ? thanks From mszlazak at aol.com Sat Mar 14 10:50:57 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 14 Mar 2009 13:50:57 -0400 Subject: [Freeswitch-users] Problem with detecting session.cause with Linksys adaptor Message-ID: <8CB72E4902A36F4-B4C-EEE@webmail-mh27.sysops.aol.com> Hello, I have an analogue line connected to SPA3102 which then sends calls to my FreeSwitch (FS) application. I can originate a call from FS and dial out the analogue line almost fine. However, an issue seems to be detecting if the PSTN line is busy. Another issue might be "hanging up" the PSTN side of the adaptor from FS side. Anyway here's the problem. If I then try originating an outbound call from FS through the pstn side of the adaptor and if a caller on the analogue line has not hung up then they hear FS dialing and any automated message that FS then sends. I can not seem to detect from FS if the PSTN line is busy for some reason with the following code. var s; while (tryCalling()) {} s.hangup(); exit(); function tryCalling() { ??? s = new Session("sofia/gateway/spa3102/14082031170 at 10.0.0.5:5061"); ??? s.waitForAnswer(30000); ??? ??? if (s.cause == "USER_BUSY") { ??? ??? return true; ??? } ??? if (s.ready()) { ??? ??? s.sleep(1000); ??? ??? s.speak("cepstral","Callie","Hello from Gino Mick Gelato"); ??? } ??? return false; } I've also tried the above with ignore_early_media=true but no luck. Thanks. Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090314/0c3aabb9/attachment.html From freeswitch at servercorps.com Sat Mar 14 10:54:34 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Sat, 14 Mar 2009 12:54:34 -0500 Subject: [Freeswitch-users] Getting a free Did number for my FS In-Reply-To: <49BBDFA8.5010805@gmail.com> References: <49BBDFA8.5010805@gmail.com> Message-ID: <92e7d2090903141054l5e090fa2vf51b423933872f22@mail.gmail.com> http://tinyurl.com/bohenh On Sat, Mar 14, 2009 at 11:47 AM, Meftah Tayeb wrote: > hello, > please ho to get a free did number ? > also, is it pocible to link it to my FS ? > thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From diego.viola at gmail.com Sat Mar 14 17:29:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 14 Mar 2009 20:29:47 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> Message-ID: <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> So how Asterisk does that "show channels" without SQL? I don't think they use SQLite internally. Just being curious. Diego On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > To clarify, -nosql turns on and off only the collecting of data for > the show commands, and can now handle higher load than the sip stack > can. ?The only thing your doing by saying -nosql is turning off the > exact functionality you say you want. ?Its similar to saying I would > like to support sip but don't want to load mod_sofia. ?There should be > no reasons to use that command anymore, if you encounter any I would > be interested in knowing what is going on. > > Mike > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > >> Yeah, but still, it would be nice to see the channels with -nosql :) >> >> I don't want to be a pain in the ass, just giving some user feedback. >> >> Regards, >> >> Diego >> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >> wrote: >>> Since we added indexes to the SQLite DB its not so bad. >>> /b >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >>> >>> Oh, I thought that SQLite wasn't that great on performance and that >>> people wanted to replace/remove it from the core. >>> >>> "On of the most interesting things about FreeSWITCH to me has been >>> the >>> fact that most data in the system such as registrations are kept in a >>> SQL database. The default installation uses SQLite internally though >>> you can easily point FreeSWITCH at one of a number of other SQL >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >>> become somewhat of a bottleneck in the core so future versions of >>> FreeSWITCH will use less of it. For example, doing a "show channels" >>> with over 500 channels in use starts to show issues. While I'm sad to >>> see SQLite go in these cases, I am anxious to see how Minessale >>> replaces it." >>> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >>> >>> I was just being curious about it :-) >>> >>> Regards, >>> >>> Diego >>> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Sat Mar 14 18:13:41 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Mar 2009 20:13:41 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> Message-ID: <019974CE-9251-4CA5-B7F9-B537C3447DFC@freeswitch.org> They walk across and lock every channel while printing the info... its a flawed way of doing things. /b On Mar 14, 2009, at 7:29 PM, Diego Viola wrote: > So how Asterisk does that "show channels" without SQL? I don't think > they use SQLite internally. > > Just being curious. > > Diego From anthony.minessale at gmail.com Sun Mar 15 06:30:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Mar 2009 08:30:24 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> Message-ID: <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> search google for bugs related to crash and show channels http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: > So how Asterisk does that "show channels" without SQL? I don't think > they use SQLite internally. > > Just being curious. > > Diego > > On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > > To clarify, -nosql turns on and off only the collecting of data for > > the show commands, and can now handle higher load than the sip stack > > can. The only thing your doing by saying -nosql is turning off the > > exact functionality you say you want. Its similar to saying I would > > like to support sip but don't want to load mod_sofia. There should be > > no reasons to use that command anymore, if you encounter any I would > > be interested in knowing what is going on. > > > > Mike > > > > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > > > >> Yeah, but still, it would be nice to see the channels with -nosql :) > >> > >> I don't want to be a pain in the ass, just giving some user feedback. > >> > >> Regards, > >> > >> Diego > >> > >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West > >> wrote: > >>> Since we added indexes to the SQLite DB its not so bad. > >>> /b > >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > >>> > >>> Oh, I thought that SQLite wasn't that great on performance and that > >>> people wanted to replace/remove it from the core. > >>> > >>> "On of the most interesting things about FreeSWITCH to me has been > >>> the > >>> fact that most data in the system such as registrations are kept in a > >>> SQL database. The default installation uses SQLite internally though > >>> you can easily point FreeSWITCH at one of a number of other SQL > >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has > >>> become somewhat of a bottleneck in the core so future versions of > >>> FreeSWITCH will use less of it. For example, doing a "show channels" > >>> with over 500 channels in use starts to show issues. While I'm sad to > >>> see SQLite go in these cases, I am anxious to see how Minessale > >>> replaces it." > >>> > >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > >>> > >>> I was just being curious about it :-) > >>> > >>> Regards, > >>> > >>> Diego > >>> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090315/de721507/attachment.html From diego.viola at gmail.com Sun Mar 15 06:56:59 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 15 Mar 2009 09:56:59 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> Message-ID: <86a32abc0903150656w5d9f8a49t233e0cdda58482@mail.gmail.com> Wow, that sucks. It's clear now why it's done this way, keep up the great work. Diego On Sun, Mar 15, 2009 at 9:30 AM, Anthony Minessale wrote: > search google for bugs related to crash and show channels > http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a > > > On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: >> >> So how Asterisk does that "show channels" without SQL? I don't think >> they use SQLite internally. >> >> Just being curious. >> >> Diego >> >> On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: >> > To clarify, -nosql turns on and off only the collecting of data for >> > the show commands, and can now handle higher load than the sip stack >> > can. ?The only thing your doing by saying -nosql is turning off the >> > exact functionality you say you want. ?Its similar to saying I would >> > like to support sip but don't want to load mod_sofia. ?There should be >> > no reasons to use that command anymore, if you encounter any I would >> > be interested in knowing what is going on. >> > >> > Mike >> > >> > >> > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: >> > >> >> Yeah, but still, it would be nice to see the channels with -nosql :) >> >> >> >> I don't want to be a pain in the ass, just giving some user feedback. >> >> >> >> Regards, >> >> >> >> Diego >> >> >> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >> >> wrote: >> >>> Since we added indexes to the SQLite DB its not so bad. >> >>> /b >> >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >> >>> >> >>> Oh, I thought that SQLite wasn't that great on performance and that >> >>> people wanted to replace/remove it from the core. >> >>> >> >>> "On of the most interesting things about FreeSWITCH to me has been >> >>> the >> >>> fact that most data in the system such as registrations are kept in a >> >>> SQL database. The default installation uses SQLite internally though >> >>> you can easily point FreeSWITCH at one of a number of other SQL >> >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >> >>> become somewhat of a bottleneck in the core so future versions of >> >>> FreeSWITCH will use less of it. For example, doing a "show channels" >> >>> with over 500 channels in use starts to show issues. While I'm sad to >> >>> see SQLite go in these cases, I am anxious to see how Minessale >> >>> replaces it." >> >>> >> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >> >>> >> >>> I was just being curious about it :-) >> >>> >> >>> Regards, >> >>> >> >>> Diego >> >>> >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Sun Mar 15 07:16:05 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 15 Mar 2009 10:16:05 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903150656w5d9f8a49t233e0cdda58482@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> <86a32abc0903150656w5d9f8a49t233e0cdda58482@mail.gmail.com> Message-ID: <86a32abc0903150716t6436e51kcf35f2c7a526df05@mail.gmail.com> Forget this, I don't want show channels without SQL anymore. Diego On Sun, Mar 15, 2009 at 9:56 AM, Diego Viola wrote: > Wow, that sucks. > > It's clear now why it's done this way, keep up the great work. > > Diego > > On Sun, Mar 15, 2009 at 9:30 AM, Anthony Minessale > wrote: >> search google for bugs related to crash and show channels >> http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a >> >> >> On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: >>> >>> So how Asterisk does that "show channels" without SQL? I don't think >>> they use SQLite internally. >>> >>> Just being curious. >>> >>> Diego >>> >>> On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: >>> > To clarify, -nosql turns on and off only the collecting of data for >>> > the show commands, and can now handle higher load than the sip stack >>> > can. ?The only thing your doing by saying -nosql is turning off the >>> > exact functionality you say you want. ?Its similar to saying I would >>> > like to support sip but don't want to load mod_sofia. ?There should be >>> > no reasons to use that command anymore, if you encounter any I would >>> > be interested in knowing what is going on. >>> > >>> > Mike >>> > >>> > >>> > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: >>> > >>> >> Yeah, but still, it would be nice to see the channels with -nosql :) >>> >> >>> >> I don't want to be a pain in the ass, just giving some user feedback. >>> >> >>> >> Regards, >>> >> >>> >> Diego >>> >> >>> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >>> >> wrote: >>> >>> Since we added indexes to the SQLite DB its not so bad. >>> >>> /b >>> >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >>> >>> >>> >>> Oh, I thought that SQLite wasn't that great on performance and that >>> >>> people wanted to replace/remove it from the core. >>> >>> >>> >>> "On of the most interesting things about FreeSWITCH to me has been >>> >>> the >>> >>> fact that most data in the system such as registrations are kept in a >>> >>> SQL database. The default installation uses SQLite internally though >>> >>> you can easily point FreeSWITCH at one of a number of other SQL >>> >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >>> >>> become somewhat of a bottleneck in the core so future versions of >>> >>> FreeSWITCH will use less of it. For example, doing a "show channels" >>> >>> with over 500 channels in use starts to show issues. While I'm sad to >>> >>> see SQLite go in these cases, I am anxious to see how Minessale >>> >>> replaces it." >>> >>> >>> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >>> >>> >>> >>> I was just being curious about it :-) >>> >>> >>> >>> Regards, >>> >>> >>> >>> Diego >>> >>> >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From marc at kasteris.com Sun Mar 15 10:19:13 2009 From: marc at kasteris.com (Marc Orenberg) Date: Sun, 15 Mar 2009 10:19:13 -0700 (PDT) Subject: [Freeswitch-users] Problems with mp3 file formats in mod_shout Message-ID: <276762.70160.qm@web50806.mail.re2.yahoo.com> Hi, I'm using mod_shout to play mp3 files, and I'm having?some trouble. I have two sets of mp3 files, one from a professional voice-prompt service, and one set that I converted from .wav files.? In both sets, all of the files play fine in Windows Media player and other players, but in both sets, there are some files which mod_shout is complaining about. It gives me errors like this: ??? ??? Note: Illegal Audio-MPEG-Header 0x41504554 at offset 0x40fa. ??? ??? Note: Trying to resync... ??? ??? Note: Hit end of (available) data during resync. When it gets these errors, it often stops playing the file before it is completed. I've tried checking / re-encoding these files using Audacity, vbrfix, mp3validator and mp3gain, and I continue to have the same problem, which is making me wonder if it's really a formating problem after all, or maybe some bug in mod_shout. I'm wondering if anybody has had these problems before, and if they have any insight into what could be going on. I'd really appreciate any help or advice. By the way, I'm using FreeSwitch version 1.0.3 on CentOS linux. Thanks in advance, Marc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090315/68619a88/attachment-0001.html From gcd at i.ph Sun Mar 15 16:46:09 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 16 Mar 2009 07:46:09 +0800 Subject: [Freeswitch-users] Getting a free Did number for my FS In-Reply-To: <49BBDFA8.5010805@gmail.com> References: <49BBDFA8.5010805@gmail.com> Message-ID: <7d0bfd8c0903151646p47700a12v1e9c601518fc8460@mail.gmail.com> perhaps you're referring to VPN (Virtual Phone Number). you can visit http://www.ipkall.com that offers free Washington state numbers. On Sun, Mar 15, 2009 at 12:47 AM, Meftah Tayeb wrote: > hello, > please ho to get a free did number ? > also, is it pocible to link it to my FS ? > thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/c51eca85/attachment.html From nik.middleton at noblesolutions.co.uk Sun Mar 15 17:01:59 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 16 Mar 2009 00:01:59 -0000 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at thesame time have all info available on realtime/runtime In-Reply-To: <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com><86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com><86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com><25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca><86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com><984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org><86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com><86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> Message-ID: To be fair, most of these messages are 4-5 years old. That said to date, I can crash * by repeatedly doing a 'show channels'. All the same FS should be robust enough to suffer this abuse. If it's not,. the issue needs to be investigated. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 15 March 2009 13:30 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but at thesame time have all info available on realtime/runtime search google for bugs related to crash and show channels http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe= utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: So how Asterisk does that "show channels" without SQL? I don't think they use SQLite internally. Just being curious. Diego On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > To clarify, -nosql turns on and off only the collecting of data for > the show commands, and can now handle higher load than the sip stack > can. The only thing your doing by saying -nosql is turning off the > exact functionality you say you want. Its similar to saying I would > like to support sip but don't want to load mod_sofia. There should be > no reasons to use that command anymore, if you encounter any I would > be interested in knowing what is going on. > > Mike > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > >> Yeah, but still, it would be nice to see the channels with -nosql :) >> >> I don't want to be a pain in the ass, just giving some user feedback. >> >> Regards, >> >> Diego >> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >> wrote: >>> Since we added indexes to the SQLite DB its not so bad. >>> /b >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >>> >>> Oh, I thought that SQLite wasn't that great on performance and that >>> people wanted to replace/remove it from the core. >>> >>> "On of the most interesting things about FreeSWITCH to me has been >>> the >>> fact that most data in the system such as registrations are kept in a >>> SQL database. The default installation uses SQLite internally though >>> you can easily point FreeSWITCH at one of a number of other SQL >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >>> become somewhat of a bottleneck in the core so future versions of >>> FreeSWITCH will use less of it. For example, doing a "show channels" >>> with over 500 channels in use starts to show issues. While I'm sad to >>> see SQLite go in these cases, I am anxious to see how Minessale >>> replaces it." >>> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >>> >>> I was just being curious about it :-) >>> >>> Regards, >>> >>> Diego >>> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/ca3a1b6b/attachment.html From anthony.minessale at gmail.com Sun Mar 15 17:15:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Mar 2009 19:15:47 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at thesame time have all info available on realtime/runtime In-Reply-To: References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> Message-ID: <191c3a030903151715w34dc4d2el82331e77f3b558d6@mail.gmail.com> Perhaps a misunderstanding? We do not suffer from any problem at all regarding "show channels". The reason for the link was to demonstrate the issue we are familiar with from our asterisk days (3-4 years ago) and to help explain how we solved it by storing the calls states in a separate table to avoid locking the channels. You can type show channel in FS all you want and all you are doing is selecting from the SQLite db. The Original question by the poster was if we can find a way to turn off SQL and still allow show channels to work and the answer is, sorry no. On Sun, Mar 15, 2009 at 7:01 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > To be fair, most of these messages are 4-5 years old. That said to date, > I can crash * by repeatedly doing a ?show channels?. All the same FS should > be robust enough to suffer this abuse. If it?s not,. the issue needs to be > investigated. > > > > > > Regards, > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 15 March 2009 13:30 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Start FreeSWITCH without any SQL but at > thesame time have all info available on realtime/runtime > > > > search google for bugs related to crash and show channels > > http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a > > On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola > wrote: > > So how Asterisk does that "show channels" without SQL? I don't think > they use SQLite internally. > > Just being curious. > > Diego > > > On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > > To clarify, -nosql turns on and off only the collecting of data for > > the show commands, and can now handle higher load than the sip stack > > can. The only thing your doing by saying -nosql is turning off the > > exact functionality you say you want. Its similar to saying I would > > like to support sip but don't want to load mod_sofia. There should be > > no reasons to use that command anymore, if you encounter any I would > > be interested in knowing what is going on. > > > > Mike > > > > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > > > >> Yeah, but still, it would be nice to see the channels with -nosql :) > >> > >> I don't want to be a pain in the ass, just giving some user feedback. > >> > >> Regards, > >> > >> Diego > >> > >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West > >> wrote: > >>> Since we added indexes to the SQLite DB its not so bad. > >>> /b > >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > >>> > >>> Oh, I thought that SQLite wasn't that great on performance and that > >>> people wanted to replace/remove it from the core. > >>> > >>> "On of the most interesting things about FreeSWITCH to me has been > >>> the > >>> fact that most data in the system such as registrations are kept in a > >>> SQL database. The default installation uses SQLite internally though > >>> you can easily point FreeSWITCH at one of a number of other SQL > >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has > >>> become somewhat of a bottleneck in the core so future versions of > >>> FreeSWITCH will use less of it. For example, doing a "show channels" > >>> with over 500 channels in use starts to show issues. While I'm sad to > >>> see SQLite go in these cases, I am anxious to see how Minessale > >>> replaces it." > >>> > >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > >>> > >>> I was just being curious about it :-) > >>> > >>> Regards, > >>> > >>> Diego > >>> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090315/d09cb47e/attachment-0001.html From shanwlin at gmail.com Mon Mar 16 00:18:00 2009 From: shanwlin at gmail.com (shawn lin) Date: Mon, 16 Mar 2009 15:18:00 +0800 Subject: [Freeswitch-users] How can I change the timeout config? Message-ID: <9fcf45ed0903160018wad781f4k99e472b5170eb7f8@mail.gmail.com> Hi all, I met a problem which I thought is caused by the timeout config setting. I use SIPp UAC <-> FreeSWITCH <-> SIPp UAS to do my test. Commands: FS: ulimit -s 256; sudo ./freeswitch UAS: ulimit -s 256; sudo sipp -sn uas -i 10.67.7.224 10.67.7.46 -trace_err UAC: ulimit -s 256; sudo sipp -sn uac -i 10.67.7.213 10.67.7.29 -trace_err -l 5000 -m 5000 -d 300000 -r 5 Infr: UAC(10.67.7.213) <-> FS internal(10.67.7.29) FS external(10.67.7.46) <-> UAS 10.67.7.224 ********************** *** My problem **** ********************** I delayed uac for 300 seconds to send the BYE messages to uas, but after about 40 seconds of pausing, uas received BYE messages from FreeSWITCH: 2009-03-16 14:36:16:274 1237185376.274928: Aborting call on an unexpected BYE for call: 1-16380 at 10.67.7.213. **** Help *** Can anyone tell me how to release the timeout limit? Best Regards! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/7e26bd25/attachment.html From nik.middleton at noblesolutions.co.uk Mon Mar 16 04:52:59 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 16 Mar 2009 11:52:59 -0000 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but atthesame time have all info available on realtime/runtime In-Reply-To: <191c3a030903151715w34dc4d2el82331e77f3b558d6@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com><86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com><25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca><86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com><984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org><86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com><86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com><191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> <191c3a030903151715w34dc4d2el82331e77f3b558d6@mail.gmail.com> Message-ID: Yup, my mistake, got the wrong end of the stick. However while we're on the subject of show channels, is it possible to get a formatted response back from the event socket? It would be nice to interrogate an event style response. Perhaps even including some other goodies such as load etc. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 16 March 2009 00:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but atthesame time have all info available on realtime/runtime Perhaps a misunderstanding? We do not suffer from any problem at all regarding "show channels". The reason for the link was to demonstrate the issue we are familiar with from our asterisk days (3-4 years ago) and to help explain how we solved it by storing the calls states in a separate table to avoid locking the channels. You can type show channel in FS all you want and all you are doing is selecting from the SQLite db. The Original question by the poster was if we can find a way to turn off SQL and still allow show channels to work and the answer is, sorry no. On Sun, Mar 15, 2009 at 7:01 PM, Nik Middleton wrote: To be fair, most of these messages are 4-5 years old. That said to date, I can crash * by repeatedly doing a 'show channels'. All the same FS should be robust enough to suffer this abuse. If it's not,. the issue needs to be investigated. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 15 March 2009 13:30 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but at thesame time have all info available on realtime/runtime search google for bugs related to crash and show channels http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe= utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: So how Asterisk does that "show channels" without SQL? I don't think they use SQLite internally. Just being curious. Diego On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > To clarify, -nosql turns on and off only the collecting of data for > the show commands, and can now handle higher load than the sip stack > can. The only thing your doing by saying -nosql is turning off the > exact functionality you say you want. Its similar to saying I would > like to support sip but don't want to load mod_sofia. There should be > no reasons to use that command anymore, if you encounter any I would > be interested in knowing what is going on. > > Mike > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > >> Yeah, but still, it would be nice to see the channels with -nosql :) >> >> I don't want to be a pain in the ass, just giving some user feedback. >> >> Regards, >> >> Diego >> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >> wrote: >>> Since we added indexes to the SQLite DB its not so bad. >>> /b >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >>> >>> Oh, I thought that SQLite wasn't that great on performance and that >>> people wanted to replace/remove it from the core. >>> >>> "On of the most interesting things about FreeSWITCH to me has been >>> the >>> fact that most data in the system such as registrations are kept in a >>> SQL database. The default installation uses SQLite internally though >>> you can easily point FreeSWITCH at one of a number of other SQL >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >>> become somewhat of a bottleneck in the core so future versions of >>> FreeSWITCH will use less of it. For example, doing a "show channels" >>> with over 500 channels in use starts to show issues. While I'm sad to >>> see SQLite go in these cases, I am anxious to see how Minessale >>> replaces it." >>> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >>> >>> I was just being curious about it :-) >>> >>> Regards, >>> >>> Diego >>> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d528c13c/attachment.html From mrene_lists at avgs.ca Mon Mar 16 04:55:14 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 16 Mar 2009 07:55:14 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but atthesame time have all info available on realtime/runtime In-Reply-To: References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com><86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com><25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca><86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com><984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org><86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com><86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com><191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> <191c3a030903151715w34dc4d2el82331e77f3b558d6@mail.gmail.com> Message-ID: <9256965F-58D6-48E1-A72E-31F13687FE26@avgs.ca> Depending on the API you're calling, of course show channels as xml show calls as xml uuid_dump [uuid] xml sofia xmlstatus and probably more than that. Math On 16-Mar-09, at 7:52 AM, Nik Middleton wrote: > Yup, my mistake, got the wrong end of the stick. > > However while we?re on the subject of show channels, is it possible > to get a formatted response back from the event socket? It would be > nice to interrogate an event style response. Perhaps even including > some other goodies such as load etc. > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Anthony Minessale > Sent: 16 March 2009 00:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but > atthesame time have all info available on realtime/runtime > > Perhaps a misunderstanding? > > We do not suffer from any problem at all regarding "show channels". > > The reason for the link was to demonstrate the issue we are familiar > with from our asterisk days (3-4 years ago) and to help explain > how we solved it by storing the calls states in a separate table to > avoid locking the channels. > > You can type show channel in FS all you want and all you are doing > is selecting from the SQLite db. > > The Original question by the poster was if we can find a way to turn > off SQL and still allow show channels to work and the > answer is, sorry no. > > > On Sun, Mar 15, 2009 at 7:01 PM, Nik Middleton > wrote: > To be fair, most of these messages are 4-5 years old. That said to > date, I can crash * by repeatedly doing a ?show channels?. All the > same FS should be robust enough to suffer this abuse. If it?s not,. > the issue needs to be investigated. > > > > > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Anthony Minessale > Sent: 15 March 2009 13:30 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but > at thesame time have all info available on realtime/runtime > > > > search google for bugs related to crash and show channels > http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a > > On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola > wrote: > > So how Asterisk does that "show channels" without SQL? I don't think > they use SQLite internally. > > Just being curious. > > Diego > > > On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris > wrote: > > To clarify, -nosql turns on and off only the collecting of data for > > the show commands, and can now handle higher load than the sip stack > > can. The only thing your doing by saying -nosql is turning off the > > exact functionality you say you want. Its similar to saying I would > > like to support sip but don't want to load mod_sofia. There > should be > > no reasons to use that command anymore, if you encounter any I would > > be interested in knowing what is going on. > > > > Mike > > > > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > > > >> Yeah, but still, it would be nice to see the channels with - > nosql :) > >> > >> I don't want to be a pain in the ass, just giving some user > feedback. > >> > >> Regards, > >> > >> Diego > >> > >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West > >> wrote: > >>> Since we added indexes to the SQLite DB its not so bad. > >>> /b > >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > >>> > >>> Oh, I thought that SQLite wasn't that great on performance and > that > >>> people wanted to replace/remove it from the core. > >>> > >>> "On of the most interesting things about FreeSWITCH to me has been > >>> the > >>> fact that most data in the system such as registrations are kept > in a > >>> SQL database. The default installation uses SQLite internally > though > >>> you can easily point FreeSWITCH at one of a number of other SQL > >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite > has > >>> become somewhat of a bottleneck in the core so future versions of > >>> FreeSWITCH will use less of it. For example, doing a "show > channels" > >>> with over 500 channels in use starts to show issues. While I'm > sad to > >>> see SQLite go in these cases, I am anxious to see how Minessale > >>> replaces it." > >>> > >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > >>> > >>> I was just being curious about it :-) > >>> > >>> Regards, > >>> > >>> Diego > >>> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/701f2305/attachment-0001.html From keithl at voxtelecom.co.za Mon Mar 16 06:36:01 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Mon, 16 Mar 2009 15:36:01 +0200 Subject: [Freeswitch-users] Core Dump on receiving a call from device with 'broken' G.722 codec. Message-ID: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> Hi, I am on fs 1.0.trunk (12530M) testing G.722 and found that when using a 'broken' configuration from a softphone configured for G.722, I get the warning on the cli: "We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. " when fs gets the invite, but then does a core dump when it tries to: Below are some of the traces and info output from before the core dump happens. I see this when I run gdb on the dumpfile. #0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/8154%172.16.1.3", timelimit_sec=30, table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=) at src/switch_ivr_originate.c:1609 1609 if (switch_core_codec_init(&write_codec, (gdb) frame 1 #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so I wonder if anybody else has seen this behavior? This happens when the destination phone is also G.722 capable (policom). If I change the "frame per packet" setting in the softphone to 2 - All works OK (but the default is 1 - so cant risk allowing G.722 if it's going to core dump fs if a user make a wrong configuration) Best Regards Keith ************************************************************************ ************************************************************* 2009-03-16 14:37:58 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/sprof1/27879998182 at 196.99.88.77 [47e0c972-1227-11de-8b8e-1789e43c417d] <.....> 2009-03-16 14:37:58 [INFO] mod_sofia.c:1310 sofia_receive_message() Asked to send early media by sofia/sprof1/27879998182 at 196.99.88.77 2009-03-16 14:37:58 [NOTICE] sofia_glue.c:2245 sofia_glue_tech_media() Pre-Answer sofia/sprof1/27879998182 at 196.99.88.77! 2009-03-16 14:37:58 [INFO] mod_sofia.c:1351 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1237190146 1237190147 IN IP4 196.99.88.77 s=FreeSWITCH c=IN IP4 196.99.88.77 t=0 0 m=audio 16932 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-03-16 14:37:58 [INFO] switch_rtp.c:1441 rtp_common_read() Auto Changing port from 172.16.0.63:29081 to 196.22.33.44:10634 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() -- checktalktime.js -- <.. In this js I do http call to collect maximum talktime allowed ..> 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() schedparms=+3600 tbhangupwarn XML hangupwarn 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182 at 196.99.88.77 to XML[27879998154 at e164route] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->27879998154 in context e164route 2009-03-16 14:37:59 [INFO] mod_dptools.c:945 info_function() CHANNEL_DATA: Event-Name: [CHANNEL_DATA] Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] FreeSWITCH-Hostname: [myfsbox] FreeSWITCH-IPv4: [196.99.88.77] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-03-16 14:37:59] Event-Date-GMT: [Mon, 16 Mar 2009 12:37:59 GMT] Event-Date-Timestamp: [1237207079387869] Event-Calling-File: [mod_dptools.c] Event-Calling-Function: [info_function] Event-Calling-Line-Number: [941] Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/sprof1/27879998182 at 196.99.88.77] Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [early] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Caller-Username: [27879998182] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MeMe] Caller-Caller-ID-Number: [27879998182] Caller-Network-Addr: [196.22.33.44] Caller-Destination-Number: [27879998154] Caller-Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Caller-Source: [mod_sofia] Caller-Context: [e164route] Caller-RDNIS: [27879998154] Caller-Channel-Name: [sofia/sprof1/27879998182 at 196.99.88.77] Caller-Profile-Index: [4] Caller-Profile-Created-Time: [1237207079387869] Caller-Channel-Created-Time: [1237207078659653] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [1237207078679638] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [196.22.33.44] variable_sip_received_port: [36745] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_Event-Name: [REQUEST_PARAMS] variable_Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] variable_FreeSWITCH-Hostname: [myfsbox] variable_FreeSWITCH-IPv4: [196.99.88.77] variable_FreeSWITCH-IPv6: [::1] variable_Event-Date-Local: [2009-03-16 14:37:58] variable_Event-Date-GMT: [Mon, 16 Mar 2009 12:37:58 GMT] variable_Event-Date-Timestamp: [1237207078659653] variable_Event-Calling-File: [sofia_reg.c] variable_Event-Calling-Function: [sofia_reg_parse_auth] variable_Event-Calling-Line-Number: [1727] variable_sip_mailbox: [879998182] variable_sip_auth_username: [27879998182] variable_sip_auth_realm: [196.99.88.77] variable_mailbox: [879998182] variable_user_name: [27879998182] variable_domain_name: [196.99.88.77] variable_record_stereo: [true] variable_default_gateway: [verso] variable_default_areacode: [87] variable_transfer_fallback_extension: [operator] variable_sip-force-expires: [180] variable_toll_allow: [domestic,international] variable_accountcode: [tbaaaa] variable_user_context: [sprof1] variable_effective_caller_id_name: [TickyBox 99999 Phone 99 Test] variable_effective_caller_id_number: [879998182] variable_outbound_caller_id_name: [879998150] variable_outbound_caller_id_number: [879998150] variable_sip_from_user: [27879998182] variable_sip_from_uri: [27879998182 at 196.99.88.77] variable_sip_from_host: [196.99.88.77] variable_sip_from_user_stripped: [27879998182] variable_sip_from_tag: [196.99.88.77] variable_sofia_profile_name: [sprof1] variable_sofia_profile_domain_name: [196.99.88.77] variable_sip_req_user: [0879998154] variable_sip_req_uri: [0879998154 at 196.99.88.77] variable_sip_req_host: [196.99.88.77] variable_sip_to_user: [0879998154] variable_sip_to_uri: [0879998154 at 196.99.88.77] variable_sip_to_host: [196.99.88.77] variable_sip_contact_user: [27879998182] variable_sip_contact_port: [22034] variable_sip_contact_uri: [27879998182 at 172.16.0.63:22034] variable_sip_contact_host: [172.16.0.63] variable_channel_name: [sofia/sprof1/27879998182 at 196.99.88.77] variable_sip_call_id: [xr125298731411533c30039109e1921f at 192.168.10.1] variable_sip_user_agent: [BrokenPhone/1.4.2] variable_sip_via_host: [172.16.0.63] variable_sip_via_port: [22034] variable_sip_via_rport: [36745] variable_presence_id: [27879998182 at 196.99.88.77] variable_switch_r_sdp: [v=0 o=2787999818 2265 2267 IN IP4 172.16.0.63 s=Broken c=IN IP4 172.16.0.63 t=0 0 m=audio 29081 RTP/AVP 9 18 101 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=candidates:-1564465265,172.16.0.63:29081,192.168.10.1:29081,192.168.20 .1:29081 ] variable_outboundcontext: [setupprepaycall] variable_remote_media_ip: [172.16.0.63] variable_remote_media_port: [29081] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_local_media_ip: [196.99.88.77] variable_local_media_port: [16932] variable_endpoint_disposition: [EARLY MEDIA] variable_sip_nat_detected: [true] variable_api_hangup_hook: [jsapi::completecall.js] variable_talktime: [6870] variable_action: [allow] variable_status: [allowed] variable_integer: [102] variable_fraction: [85] variable_saytalktime: [60:0] variable_schedparms: [+3600 tbhangupwarn XML hangupwarn] variable_bridgejscb: [{api_hangup_hook=jsapi::completecall.js}] variable_max_forwards: [67] variable_current_application: [info] 2009-03-16 14:37:59 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-03-16 14:37:59 [WARNING] switch_core_codec.c:499 switch_core_codec_init() Codec G722 Exists but not at the desired implementation. 8000hz 820ms 2009-03-16 14:37:59 [ERR] sofia_glue.c:1700 sofia_glue_tech_set_codec() Can't load codec? 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/sprof1/27879998182 at 196.99.88.77 has no read codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:585 switch_core_session_write_frame() sofia/sprof1/27879998182 at 196.99.88.77 has no write codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/sprof1/27879998182 at 196.99.88.77 has no read codec. 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182 at 196.99.88.77 to XML[8154 at toregext] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->8154 in context toregext 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/27879998182.2009-03-16- 14-37-59.wav 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features <... output from info application ...> 2009-03-16 14:38:00 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/voxwan/8154 [48c2f400-1227-11de-8b8e-1789e43c417d] Segmentation fault (core dumped) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/7b37f236/attachment-0001.html From mrene_lists at avgs.ca Mon Mar 16 06:46:08 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 16 Mar 2009 09:46:08 -0400 Subject: [Freeswitch-users] Core Dump on receiving a call from device with 'broken' G.722 codec. In-Reply-To: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> Message-ID: Hi, You should be reporting this on JIRA ( see http://wiki.freeswitch.org/wiki/Reporting_Bugs ) Also please include a "bt", not just "frame 1" as it doesnt give out much information. Math On 16-Mar-09, at 9:36 AM, Keith Laaks wrote: > Hi, > > I am on fs 1.0.trunk (12530M) testing G.722 and found that when > using a ?broken? configuration from a softphone configured for G. > 722, I get the warning on the cli: > > ?We were told to use ptime 20 but what they meant to say was 820 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sipura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who knows what will happen.. > ? > when fs gets the invite, but then does a core dump when it tries to: > > > > > Below are some of the traces and info output from before the core > dump happens. > > I see this when I run gdb on the dumpfile. > > #0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, > bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/ > 8154%172.16.1.3", timelimit_sec=30, > table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, > caller_profile_override=0x0, ovars=0x0, flags=) > at src/switch_ivr_originate.c:1609 > 1609 if > (switch_core_codec_init(&write_codec, > > (gdb) frame 1 > #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so > > > I wonder if anybody else has seen this behavior? > > This happens when the destination phone is also G.722 capable > (policom). > If I change the ?frame per packet? setting in the softphone to 2 ? > All works OK (but the default is 1 ? so cant risk allowing G.722 if > it?s going to core dump fs if a user make a wrong configuration) > > > Best Regards > > Keith > > > ************************************************************************************************************************************* > > > 2009-03-16 14:37:58 [NOTICE] switch_channel.c:592 > switch_channel_set_name() New Channel sofia/sprof1/27879998182 at 196.99.88.77 > [47e0c972-1227-11de-8b8e-1789e43c417d] > > <.....> > > 2009-03-16 14:37:58 [INFO] mod_sofia.c:1310 sofia_receive_message() > Asked to send early media by sofia/sprof1/27879998182 at 196.99.88.77 > 2009-03-16 14:37:58 [NOTICE] sofia_glue.c:2245 > sofia_glue_tech_media() Pre-Answer sofia/sprof1/27879998182 at 196.99.88.77 > ! > 2009-03-16 14:37:58 [INFO] mod_sofia.c:1351 sofia_receive_message() > Ring SDP: > v=0 > o=FreeSWITCH 1237190146 1237190147 IN IP4 196.99.88.77 > s=FreeSWITCH > c=IN IP4 196.99.88.77 > t=0 0 > m=audio 16932 RTP/AVP 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2009-03-16 14:37:58 [INFO] switch_rtp.c:1441 rtp_common_read() Auto > Changing port from 172.16.0.63:29081 to 196.22.33.44:10634 > > 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() -- > checktalktime.js -- > > <.. In this js I do http call to collect maximum talktime allowed ..> > > 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() > schedparms=+3600 tbhangupwarn XML hangupwarn > 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 > switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182 at 196.99.88.77 > to XML[27879998154 at e164route] > 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing MeMe->27879998154 in context e164route > 2009-03-16 14:37:59 [INFO] mod_dptools.c:945 info_function() > CHANNEL_DATA: > Event-Name: [CHANNEL_DATA] > Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] > FreeSWITCH-Hostname: [myfsbox] > FreeSWITCH-IPv4: [196.99.88.77] > FreeSWITCH-IPv6: [::1] > Event-Date-Local: [2009-03-16 14:37:59] > Event-Date-GMT: [Mon, 16 Mar 2009 12:37:59 GMT] > Event-Date-Timestamp: [1237207079387869] > Event-Calling-File: [mod_dptools.c] > Event-Calling-Function: [info_function] > Event-Calling-Line-Number: [941] > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/sprof1/27879998182 at 196.99.88.77] > Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [early] > Channel-Read-Codec-Name: [G722] > Channel-Read-Codec-Rate: [16000] > Channel-Write-Codec-Name: [G722] > Channel-Write-Codec-Rate: [16000] > Caller-Username: [27879998182] > Caller-Dialplan: [XML] > Caller-Caller-ID-Name: [MeMe] > Caller-Caller-ID-Number: [27879998182] > Caller-Network-Addr: [196.22.33.44] > Caller-Destination-Number: [27879998154] > Caller-Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] > Caller-Source: [mod_sofia] > Caller-Context: [e164route] > Caller-RDNIS: [27879998154] > Caller-Channel-Name: [sofia/sprof1/27879998182 at 196.99.88.77] > Caller-Profile-Index: [4] > Caller-Profile-Created-Time: [1237207079387869] > Caller-Channel-Created-Time: [1237207078659653] > Caller-Channel-Answered-Time: [0] > Caller-Channel-Progress-Time: [0] > Caller-Channel-Progress-Media-Time: [1237207078679638] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > variable_sip_received_ip: [196.22.33.44] > variable_sip_received_port: [36745] > variable_sip_via_protocol: [udp] > variable_sip_authorized: [true] > variable_Event-Name: [REQUEST_PARAMS] > variable_Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] > variable_FreeSWITCH-Hostname: [myfsbox] > variable_FreeSWITCH-IPv4: [196.99.88.77] > variable_FreeSWITCH-IPv6: [::1] > variable_Event-Date-Local: [2009-03-16 14:37:58] > variable_Event-Date-GMT: [Mon, 16 Mar 2009 12:37:58 GMT] > variable_Event-Date-Timestamp: [1237207078659653] > variable_Event-Calling-File: [sofia_reg.c] > variable_Event-Calling-Function: [sofia_reg_parse_auth] > variable_Event-Calling-Line-Number: [1727] > variable_sip_mailbox: [879998182] > variable_sip_auth_username: [27879998182] > variable_sip_auth_realm: [196.99.88.77] > variable_mailbox: [879998182] > variable_user_name: [27879998182] > variable_domain_name: [196.99.88.77] > variable_record_stereo: [true] > variable_default_gateway: [verso] > variable_default_areacode: [87] > variable_transfer_fallback_extension: [operator] > variable_sip-force-expires: [180] > variable_toll_allow: [domestic,international] > variable_accountcode: [tbaaaa] > variable_user_context: [sprof1] > variable_effective_caller_id_name: [TickyBox 99999 Phone 99 Test] > variable_effective_caller_id_number: [879998182] > variable_outbound_caller_id_name: [879998150] > variable_outbound_caller_id_number: [879998150] > variable_sip_from_user: [27879998182] > variable_sip_from_uri: [27879998182 at 196.99.88.77] > variable_sip_from_host: [196.99.88.77] > variable_sip_from_user_stripped: [27879998182] > variable_sip_from_tag: [196.99.88.77] > variable_sofia_profile_name: [sprof1] > variable_sofia_profile_domain_name: [196.99.88.77] > variable_sip_req_user: [0879998154] > variable_sip_req_uri: [0879998154 at 196.99.88.77] > variable_sip_req_host: [196.99.88.77] > variable_sip_to_user: [0879998154] > variable_sip_to_uri: [0879998154 at 196.99.88.77] > variable_sip_to_host: [196.99.88.77] > variable_sip_contact_user: [27879998182] > variable_sip_contact_port: [22034] > variable_sip_contact_uri: [27879998182 at 172.16.0.63:22034] > variable_sip_contact_host: [172.16.0.63] > variable_channel_name: [sofia/sprof1/27879998182 at 196.99.88.77] > variable_sip_call_id: [xr125298731411533c30039109e1921f at 192.168.10.1] > variable_sip_user_agent: [BrokenPhone/1.4.2] > variable_sip_via_host: [172.16.0.63] > variable_sip_via_port: [22034] > variable_sip_via_rport: [36745] > variable_presence_id: [27879998182 at 196.99.88.77] > variable_switch_r_sdp: [v=0 > o=2787999818 2265 2267 IN IP4 172.16.0.63 > s=Broken > c=IN IP4 172.16.0.63 > t=0 0 > m=audio 29081 RTP/AVP 9 18 101 > a=rtpmap:9 G722/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a > = > candidates > :-1564465265,172.16.0.63:29081,192.168.10.1:29081,192.168.20.1:29081 > ] > variable_outboundcontext: [setupprepaycall] > variable_remote_media_ip: [172.16.0.63] > variable_remote_media_port: [29081] > variable_read_codec: [G722] > variable_read_rate: [16000] > variable_write_codec: [G722] > variable_write_rate: [16000] > variable_local_media_ip: [196.99.88.77] > variable_local_media_port: [16932] > variable_endpoint_disposition: [EARLY MEDIA] > variable_sip_nat_detected: [true] > variable_api_hangup_hook: [jsapi::completecall.js] > variable_talktime: [6870] > variable_action: [allow] > variable_status: [allowed] > variable_integer: [102] > variable_fraction: [85] > variable_saytalktime: [60:0] > variable_schedparms: [+3600 tbhangupwarn XML hangupwarn] > variable_bridgejscb: [{api_hangup_hook=jsapi::completecall.js}] > variable_max_forwards: [67] > variable_current_application: [info] > > > > 2009-03-16 14:37:59 [WARNING] mod_sofia.c:739 sofia_read_frame() We > were told to use ptime 20 but what they meant to say was 820 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sipura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who knows what will happen.. > 2009-03-16 14:37:59 [WARNING] switch_core_codec.c:499 > switch_core_codec_init() Codec G722 Exists but not at the desired > implementation. 8000hz 820ms > 2009-03-16 14:37:59 [ERR] sofia_glue.c:1700 > sofia_glue_tech_set_codec() Can't load codec? > 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 > switch_core_session_read_frame() sofia/ > sprof1/27879998182 at 196.99.88.77 has no read codec. > 2009-03-16 14:37:59 [ERR] switch_core_io.c:585 > switch_core_session_write_frame() sofia/sprof1/27879998182 at 196.99.88.77 > has no write codec. > 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 > switch_core_session_read_frame() sofia/ > sprof1/27879998182 at 196.99.88.77 has no read codec. > 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 > switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182 at 196.99.88.77 > to XML[8154 at toregext] > 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing MeMe->8154 in context toregext > 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 > execute_extension::dx XML features > 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ > usr/local/freeswitch/recordings/27879998182.2009-03-16-14-37-59.wav > 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 > execute_extension::cf XML features > > > > 2009-03-16 14:38:00 [NOTICE] switch_channel.c:592 > switch_channel_set_name() New Channel sofia/voxwan/8154 > [48c2f400-1227-11de-8b8e-1789e43c417d] > Segmentation fault (core dumped) > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/de5c4fc0/attachment-0001.html From brian at freeswitch.org Mon Mar 16 06:48:03 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 08:48:03 -0500 Subject: [Freeswitch-users] Core Dump on receiving a call from device with 'broken' G.722 codec. In-Reply-To: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> Message-ID: http://wiki.freeswitch.org/wiki/Reporting_Bugs Keith, Please read the link above... open a jira and collect a sip trace of this also and "attach" it. /b On Mar 16, 2009, at 8:36 AM, Keith Laaks wrote: > Hi, > > I am on fs 1.0.trunk (12530M) testing G.722 and found that when > using a ?broken? configuration from a softphone configured for G. > 722, I get the warning on the cli: > > ?We were told to use ptime 20 but what they meant to say was 820 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sipura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who knows what will happen.. > ? > when fs gets the invite, but then does a core dump when it tries to: > > > > > Below are some of the traces and info output from before the core > dump happens. > > I see this when I run gdb on the dumpfile. > > #0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, > bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/ > 8154%172.16.1.3", timelimit_sec=30, > table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, > caller_profile_override=0x0, ovars=0x0, flags=) > at src/switch_ivr_originate.c:1609 > 1609 if > (switch_core_codec_init(&write_codec, > > (gdb) frame 1 > #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so > > > I wonder if anybody else has seen this behavior? > > This happens when the destination phone is also G.722 capable > (policom). > If I change the ?frame per packet? setting in the softphone to 2 ? > All works OK (but the default is 1 ? so cant risk allowing G.722 if > it?s going to core dump fs if a user make a wrong configuration) > > > Best Regards > > Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/a2d6e21a/attachment.html From fs at xenpad.eu Mon Mar 16 08:29:32 2009 From: fs at xenpad.eu (fs at xenpad.eu) Date: Mon, 16 Mar 2009 16:29:32 +0100 (CET) Subject: [Freeswitch-users] Problem with shortened local extensions In-Reply-To: References: Message-ID: Hi, I have a (probably dumb) question that I just spent over 5 hours on: I have a 1.0.2 version running with 6 extensions, 2 SIP trunks and gtalk OK. I tried to change the extensions from 100x to 10x (100-109 actually) I changed the info in the directory, hunted all references to patterns like 10[01][0-9], and replaced them with 10[0-9], and changed the DID routing to the new extension numbers. No errors at load time, sofia status profile internal shows all extensions registered; each extension can call it's own voicemail, and all provided examples (5000, 9998, ...) work OK. The catch is, it's not possible to call another extension, and the routing from incoming SIP trunks fail (the extension is not available, please oleave a message). I'm all out of ideas. I first tried that with a load of other changes; rolled back everything to the working setup, and just did this one change; all to no avail. What am I missing? TIA, Laurent From steve.d.ward at gmail.com Mon Mar 16 07:19:15 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 10:19:15 -0400 Subject: [Freeswitch-users] sip trunking question Message-ID: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> I'm trying to set up a sip trunk between a FS and * box, and right now I'm having trouble getting things set up so I make a call from a sip phone registered with my FS box to a sip phone registered w/ my Asterisk box. I have a gateway defined as in directory/default/example.com.xml and in my dialplan I'm trying to do a bridge w/ something like "sofia/gateway/${default_gateway}/12345." When I try to make the call I see from the console: ... New Channel sofia/external/12345 ... ... STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] ... Hangup sofia/external/12345 [CS_INIT] [DESTINATION_OUT_OF_ORDER] The Originate fails. I tried sticking to what the instructions laid out for this in the Connecting FS and Asterisk wiki page, so I'd appreciate some help in figuring out what's going on. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/6ba86a85/attachment.html From kerrada2003 at yahoo.com Mon Mar 16 08:58:40 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 16 Mar 2009 08:58:40 -0700 (PDT) Subject: [Freeswitch-users] FS Database Message-ID: <135187.79162.qm@web33708.mail.mud.yahoo.com> Hi, Is it possible to access FS DB to retrieve data? Where can i find details about that? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/1ffeac71/attachment.html From msc at freeswitch.org Mon Mar 16 08:59:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 08:59:18 -0700 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> Message-ID: <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> 2009/3/16 Steven Ward : > I'm trying to set up a sip trunk between a FS and * box, and right now I'm > having trouble getting things set up so I make a call from a sip phone > registered with my FS box to a sip phone registered w/ my Asterisk box. > > I have a gateway defined as in directory/default/example.com.xml and in my > dialplan I'm trying to do a bridge w/ something like > "sofia/gateway/${default_gateway}/12345." > > When I try to make the call I see from the console: > > ... New Channel sofia/external/12345 ... > ... STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] > ... Hangup sofia/external/12345 [CS_INIT] [DESTINATION_OUT_OF_ORDER] What is your network setup? The gateway you created is using the external profile and trying to do a STUN lookup. Is that what you are trying to do? Just confirming. -MC From msc at freeswitch.org Mon Mar 16 09:06:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 09:06:05 -0700 Subject: [Freeswitch-users] Problem with shortened local extensions In-Reply-To: References: Message-ID: <87f2f3b90903160906q693370c8s6b68eafd528f6401@mail.gmail.com> On Mon, Mar 16, 2009 at 8:29 AM, wrote: > ? Hi, > > ? I have a (probably dumb) question that I just spent over 5 hours on: > I have a 1.0.2 version running with 6 extensions, 2 SIP trunks and gtalk OK. Ouch! Any way you could update? We are on the verge of releasing 1.0.4; 1.0.2 is OLD. :) > > ? I tried to change the extensions from 100x to 10x (100-109 actually) > ? I changed the info in the directory, hunted all references to patterns > like 10[01][0-9], and replaced them with 10[0-9], and changed the DID > routing to the new extension numbers. > > ? No errors at load time, sofia status profile internal shows all > extensions registered; each extension can call it's own voicemail, and > all provided examples (5000, 9998, ...) work OK. > > ? The catch is, it's not possible to call another extension, and the > routing from incoming SIP trunks fail (the extension is not available, please > oleave a message). Best bet here is to read up on this page: http://wiki.freeswitch.org/wiki/Reporting_Bugs That will help you do stuff like this: turn on debugging (press F8 at the CLI) then make a test call, capture output, put it into a pastebin. I'm sure it's something basic, but without seeing what's happening it's hard to diagnose. Also, pastebin your default.xml dialplan file, one or more of your directory files, like 100.xml (or whatever you named them), and finally do "sofia status profile internal" at the CLI and pastebin the output. -MC > > ? I'm all out of ideas. I first tried that with a load of other changes; > rolled back everything to the working setup, and just did this one > change; all to no avail. > > ? What am I missing? > > ? TIA, > ? ?Laurent > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Mon Mar 16 09:24:34 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Mar 2009 12:24:34 -0400 Subject: [Freeswitch-users] FS Database In-Reply-To: <135187.79162.qm@web33708.mail.mud.yahoo.com> References: <135187.79162.qm@web33708.mail.mud.yahoo.com> Message-ID: <1BF9A176-1CE8-4D53-894B-E456DA8897A0@jerris.com> Which data? On Mar 16, 2009, at 11:58 AM, Ali Al-Rubaie wrote: > Hi, > > Is it possible to access FS DB to retrieve data? Where can i find > details about that? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d0d0a1b0/attachment-0001.html From msc at freeswitch.org Mon Mar 16 09:28:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 09:28:17 -0700 Subject: [Freeswitch-users] FS Database In-Reply-To: <135187.79162.qm@web33708.mail.mud.yahoo.com> References: <135187.79162.qm@web33708.mail.mud.yahoo.com> Message-ID: <87f2f3b90903160928u54583af6r48e1ba0a7f41b43e@mail.gmail.com> > Is it possible to access FS DB to retrieve data? Where can i find details > about that? Could you be a little more specific? Also, on a standard Linux/Unix install you could check here: /usr/local/freeswitch/db/*db -MC From fs at xenpad.eu Mon Mar 16 09:36:27 2009 From: fs at xenpad.eu (fs at xenpad.eu) Date: Mon, 16 Mar 2009 17:36:27 +0100 (CET) Subject: [Freeswitch-users] Problem with shortened local extensions In-Reply-To: <87f2f3b90903160906q693370c8s6b68eafd528f6401@mail.gmail.com> References: <87f2f3b90903160906q693370c8s6b68eafd528f6401@mail.gmail.com> Message-ID: Hi, On Mon, 16 Mar 2009, Michael Collins wrote: >> ? I have a (probably dumb) question that I just spent over 5 hours on: >> I have a 1.0.2 version running with 6 extensions, 2 SIP trunks and gtalk OK. > > Ouch! Any way you could update? We are on the verge of releasing > 1.0.4; 1.0.2 is OLD. :) Then I'll skip 1.0.3 and wait for 1.0.4 ;) > That will help you do stuff like this: > turn on debugging (press F8 at the CLI) then make a test call, capture > output, put it into a pastebin. OK -- this is a live system; I worked on it this weekend. I'll try again with the short extension and collect all data. Should 1.0.4 be out in the meantime, I'll upgrade before I try. Cheers, Laurent -- +-----------------------------------------------------------------------+ | E-mail Home page http://case.jorune.net/ | | We do what we must because we can. | +-----------------------------------------------------------------------+ From kerrada2003 at yahoo.com Mon Mar 16 09:46:04 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 16 Mar 2009 09:46:04 -0700 (PDT) Subject: [Freeswitch-users] FS DB In-Reply-To: Message-ID: <595500.67715.qm@web33706.mail.mud.yahoo.com> Actually, I'm trying to explore what data is stored in the DB and how can we retrieve it, if it is posible. You may write some scripts to retrieve the data and use it in other supporting application. Message: 7 Date: Mon, 16 Mar 2009 12:24:34 -0400 From: Michael Jerris Subject: Re: [Freeswitch-users] FS Database To: freeswitch-users at lists.freeswitch.org Message-ID: <1BF9A176-1CE8-4D53-894B-E456DA8897A0 at jerris.com> Content-Type: text/plain; charset="us-ascii" Which data? On Mar 16, 2009, at 11:58 AM, Ali Al-Rubaie wrote: > Hi, > > Is it possible to access FS DB to retrieve data? Where can i find > details about that? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d0d0a1b0/attachment.html ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 33, Issue 67 ************************************************ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/84be11c8/attachment.html From Mark.Tabron at rnid-typetalk.org.uk Mon Mar 16 09:53:25 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Mon, 16 Mar 2009 16:53:25 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room. So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be. Any ideas? Pastebin debug output is in my reply below. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 13 March 2009 14:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! http://pastebin.freeswitch.org/7751 Will setup an IRC client and see if I can log onto the channel. Thanks again! -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 March 2009 16:50 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 > My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so > apologies if some of my terminology isn't quite correct. Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > > > > Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've > hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the > card is green and wanrouter is installed and up in TDM_API mode, with the > connection status showing as connected. ?Configured Openzap for 9 b and 1 d > channel as described in Freeswitch Wiki. Then created a diaplan to fire off > any calls preceded by 9 to the next available openzap channel. Looks good so far... > The problem I have is when I initiate an external call (using 9xxxxxxx) from > an extension I can see Freeswitch allocating the call to the next available > channel but then the just sits there and times out after 1 minute. With the > cause stated as ORIGINATOR_CANCEL (guessing this is the time out) okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has lots of useful information for how to gather log information, how to use the pastebin, etc. Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers: To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org). Welcome to the wonderful world of telephony debugging! -MC P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.) _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From steve.d.ward at gmail.com Mon Mar 16 10:15:25 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 13:15:25 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> Message-ID: <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> Yes, the obvious is the case. :) I don't want to do a STUN lookup - the two machines are on the same LAN. What's the best way to get the gateway to not do a STUN lookup? Do I need to disable STUN for the external profile or make this gateway use a different profile? Thanks. SW On Mon, Mar 16, 2009 at 11:59 AM, Michael Collins wrote: > 2009/3/16 Steven Ward : > > I'm trying to set up a sip trunk between a FS and * box, and right now > I'm > > having trouble getting things set up so I make a call from a sip phone > > registered with my FS box to a sip phone registered w/ my Asterisk box. > > > > I have a gateway defined as in directory/default/example.com.xml and in > my > > dialplan I'm trying to do a bridge w/ something like > > "sofia/gateway/${default_gateway}/12345." > > > > When I try to make the call I see from the console: > > > > ... New Channel sofia/external/12345 ... > > ... STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] > > ... Hangup sofia/external/12345 [CS_INIT] [DESTINATION_OUT_OF_ORDER] > > What is your network setup? The gateway you created is using the > external profile and trying to do a STUN lookup. Is that what you are > trying to do? Just confirming. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/f5936e5f/attachment.html From msc at freeswitch.org Mon Mar 16 10:15:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 10:15:53 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> Any chance you can give one of us access to this system? Best thing to do would be to join #openzap on irc.freenode.net. -MC (IRC: mercutioviz) On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron wrote: > Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room. > > So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be. > > Any ideas? Pastebin debug output is in my reply below. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron > Sent: 13 March 2009 14:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). > > Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! > > http://pastebin.freeswitch.org/7751 > > Will setup an IRC client and see if I can log onto the channel. > > Thanks again! > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 12 March 2009 16:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > >> My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so >> apologies if some of my terminology isn't quite correct. > > Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > >> >> >> >> Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've >> hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the >> card is green and wanrouter is installed and up in TDM_API mode, with the >> connection status showing as connected. ?Configured Openzap for 9 b and 1 d >> channel as described in Freeswitch Wiki. Then created a diaplan to fire off >> any calls preceded by 9 to the next available openzap channel. > > Looks good so far... > >> The problem I have is when I initiate an external call (using 9xxxxxxx) from >> an extension I can see Freeswitch allocating the call to the next available >> channel but then the just sits there and times out after 1 minute. With the >> cause stated as ORIGINATOR_CANCEL (guessing this is the time out) > > okay, some debugging info will be useful. Please read this wiki page first: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It has lots of useful information for how to gather log information, > how to use the pastebin, etc. > > Specifically for this issue you'll need to use the pastebin because > there will be so much information. Here are some pointers: > > To see what's happening with openzap you'll need to use the "oz list" > and "oz dump 1" at the command line (CLI). You'll also need to turn on > debugging so that PRI messages show up. You'll need to capture the > output on the CLI and put it into the pastebin. > (http://pastebin.freeswitch.org). > > Welcome to the wonderful world of telephony debugging! > -MC > > P.S. - We have a few IRC channels where you can join to get more > real-time support: > #freeswitch and #openzap on irc.freenode.net. (More details are in the > wiki page I mentioned above.) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > -------------------------------------------------------------------------------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > -------------------------------------------------------------------------------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Mar 16 10:24:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 10:24:15 -0700 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> Message-ID: <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> 2009/3/16 Steven Ward : > Yes, the obvious is the case.? :) I don't want to do a STUN lookup - the two > machines are on the same LAN. > > What's the best way to get the gateway to not do a STUN lookup?? Do I need > to disable STUN for the external > profile or make this gateway use a different profile? In which directory do you create your gateway file? If you created it in sip_profiles/external/ then try moving it over to sip_profiles/internal/ and see what happens... -MC P.S. - You could also disable STUN on your external profile, but since your two boxes are on the same LAN I would suggestion that the "proper" way to handle this situation is to have your gateway use the internal profile. From keithl at voxtelecom.co.za Mon Mar 16 10:25:06 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Mon, 16 Mar 2009 19:25:06 +0200 Subject: [Freeswitch-users] Module Mod Native File - How to handle multiple rates under the same codec. Message-ID: <1B99233662E2104983E3550185D3ED73497A49@xena.internal.datapro.co.za> Hi, To minimize/eliminate transcoding, I am using mod native file, with a set of transcoded prompts with the appropriate set of file extensions. Everything works as advertised when using the traditional codecs such as pcma, g729, gsm. But speex support is a bit of a challenge. Freeswitch supports: speex at 8000h@20i, speex at 16000h@20i, speex at 32000h@20i. When I setup calls using these various codec flavors, I can see using info (and of course detect by ear) that indeed the call is running at the different 'rates'. I see that regardless of which one I use, the "variable_read_codec" and "variable_write_codec" remains "SPEEX", but depending on the flavor, a "variable_read_rate" and "variable_write_rate" of either 8000,16000 or 32000. But when I try play a file when in 16000 or 32000, I get: 2009-03-15 16:31:13 [INFO] mod_native_file.c:81 native_file_file_open() Opening File [/usr/local/freeswitch/sounds/en/us/callie/all/16000/SUCCESS.SPEEX] 8000hz 2009-03-15 16:31:13 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match. I created my SPEEX files using: speexenc -w (note the -w option for 16kHz wideband) So, even though the call is setup using a wideband 16kHz codec, it appears that mod native file is expecting a 8kHz file for all the SPEEX flavors. What am I missing here? Is this module limited to 8KHz rates? I am on 1.0.trunk (12530M). I have not yet looked at these codecs: G7221 at 16000h, G7221 at 32000h, CELT at 32000h, CELT at 48000h, but as these also have multiple rates for the same codec - I expect same issue. I am using ${ variable_read_rate } in the filename path, so fs looks at a set of files encoded with a matching sample rate. But looks like it's always looking for a 8KHz file. If you have had any experience with this, please let me have your advice. Thanks Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/f4c71603/attachment.html From mrene_lists at avgs.ca Mon Mar 16 10:26:32 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 16 Mar 2009 13:26:32 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> Message-ID: <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> The reason its using stun is because your external-sip-ip and external- rtp-ip params are starting with stun: As Michael says, the external profile is meant to do nat-traversal, if you dont need it, use the internal one. Math On 16-Mar-09, at 1:24 PM, Michael Collins wrote: > 2009/3/16 Steven Ward : >> Yes, the obvious is the case. :) I don't want to do a STUN lookup >> - the two >> machines are on the same LAN. >> >> What's the best way to get the gateway to not do a STUN lookup? Do >> I need >> to disable STUN for the external >> profile or make this gateway use a different profile? > > In which directory do you create your gateway file? If you created it > in sip_profiles/external/ then try moving it over to > sip_profiles/internal/ and see what happens... > > -MC > > P.S. - You could also disable STUN on your external profile, but since > your two boxes are on the same LAN I would suggestion that the > "proper" way to handle this situation is to have your gateway use the > internal profile. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steve.d.ward at gmail.com Mon Mar 16 10:27:52 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 13:27:52 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> Message-ID: <4ea6e8f20903161027q5d2c18bfmf85c493ac3f84cf9@mail.gmail.com> Thanks. I created the gateway file in conf/directory/default/ On Mon, Mar 16, 2009 at 1:24 PM, Michael Collins wrote: > 2009/3/16 Steven Ward : > > Yes, the obvious is the case. :) I don't want to do a STUN lookup - the > two > > machines are on the same LAN. > > > > What's the best way to get the gateway to not do a STUN lookup? Do I > need > > to disable STUN for the external > > profile or make this gateway use a different profile? > > In which directory do you create your gateway file? If you created it > in sip_profiles/external/ then try moving it over to > sip_profiles/internal/ and see what happens... > > -MC > > P.S. - You could also disable STUN on your external profile, but since > your two boxes are on the same LAN I would suggestion that the > "proper" way to handle this situation is to have your gateway use the > internal profile. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/480d5bd5/attachment.html From steve.d.ward at gmail.com Mon Mar 16 10:39:32 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 13:39:32 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> Message-ID: <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> I simply moved the file defining the gateway to conf/sip_profiles/internal Well, when calling from extension 1000 to 70904, what I see on the console (debug mode) is: 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/1000 at pbx-sip-3.usa.internal.net Execute bridge(sofia/gateway/${default_gateway}/70904) 2009-03-16 13:35:39 [DEBUG] switch_core_session.c:1286 switch_core_session_exec() sofia/internal/1000 at pbx-sip-3.usa.internal.netExpanded String bridge(sofia/gateway/ pbx-sip-4.usa.internal.net/70904) 2009-03-16 13:35:39 [ERR] mod_sofia.c:2379 sofia_outgoing_channel() Invalid Gateway 2009-03-16 13:35:39 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2009-03-16 13:35:39 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2009-03-16 13:35:39 [DEBUG] switch_ivr_originate.c:2014 switch_ivr_originate() Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2009-03-16 13:35:39 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT What else am I missing and not doing right? Thanks again for your help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/18d7c427/attachment.html From brian at freeswitch.org Mon Mar 16 10:44:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 12:44:18 -0500 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> Message-ID: <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> I would almost bet your xml is wrong when you moved it.. care to share that little bit of info? /b On Mar 16, 2009, at 12:39 PM, Steven Ward wrote: > I simply moved the file defining the gateway to conf/sip_profiles/ > internal > > Well, when calling from extension 1000 to 70904, what I see on the > console (debug mode) is: > > 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1000 at pbx-sip-3.usa.internal.net > Execute bridge(sofia/gateway/${default_gateway}/70904) > 2009-03-16 13:35:39 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1000 at pbx-sip-3.usa.internal.net > Expanded String bridge(sofia/gateway/pbx-sip-4.usa.internal.net/ > 70904) > 2009-03-16 13:35:39 [ERR] mod_sofia.c:2379 sofia_outgoing_channel() > Invalid Gateway > 2009-03-16 13:35:39 [NOTICE] mod_sofia.c:2591 > sofia_outgoing_channel() Close Channel N/A [CS_NEW] > 2009-03-16 13:35:39 [ERR] switch_ivr_originate.c:1425 > switch_ivr_originate() Cannot create outgoing channel of type > [sofia] cause: [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [DEBUG] switch_ivr_originate.c:2014 > switch_ivr_originate() Originate Resulted in Error Cause: 28 > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [INFO] mod_dptools.c:1998 > audio_bridge_function() Originate Failed. Cause: > INVALID_NUMBER_FORMAT > > > What else am I missing and not doing right? Thanks again for your > help. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/4d982fb9/attachment-0001.html From msc at freeswitch.org Mon Mar 16 10:47:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 10:47:32 -0700 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> Message-ID: <87f2f3b90903161047n4dbcf8c4i6e4924e9d9729a09@mail.gmail.com> 2009/3/16 Steven Ward : > I simply moved the file defining the gateway to conf/sip_profiles/internal > > Well, when calling from extension 1000 to 70904, what I see on the console > (debug mode) is: > > 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() > sofia/internal/1000 at pbx-sip-3.usa.internal.net Execute > bridge(sofia/gateway/${default_gateway}/70904) > 2009-03-16 13:35:39 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1000 at pbx-sip-3.usa.internal.net > Expanded String bridge(sofia/gateway/pbx-sip-4.usa.internal.net/70904) > 2009-03-16 13:35:39 [ERR] mod_sofia.c:2379 sofia_outgoing_channel() Invalid > Gateway ^^^^^^^^^^^^^^^^ There's your key. Invalid gateway means just that: you're dialing a gw that doesn't exist. What is your gateway name in the file? Is it really "pbx-sip-4.usa.internal.net" ? does that resolve to an internal IP address? -MC > 2009-03-16 13:35:39 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() Close > Channel N/A [CS_NEW] > 2009-03-16 13:35:39 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() > Cannot create outgoing channel of type [sofia] cause: > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [DEBUG] switch_ivr_originate.c:2014 > switch_ivr_originate() Originate Resulted in Error Cause: 28 > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [INFO] mod_dptools.c:1998 audio_bridge_function() > Originate Failed.? Cause: INVALID_NUMBER_FORMAT > > > What else am I missing and not doing right?? Thanks again for your help. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve.d.ward at gmail.com Mon Mar 16 10:51:20 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 13:51:20 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> Message-ID: <4ea6e8f20903161051k1c6609ecp1b0ec57a68810e8@mail.gmail.com> Sure thing. Here it is: In vars.conf I supplied the variables' values: 2009/3/16 Brian West > I would almost bet your xml is wrong when you moved it.. care to share that > little bit of info? > /b > > On Mar 16, 2009, at 12:39 PM, Steven Ward wrote: > > I simply moved the file defining the gateway to > conf/sip_profiles/internal > > Well, when calling from extension 1000 to 70904, what I see on the console > (debug mode) is: > > 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() > sofia/internal/1000 at pbx-sip-3.usa.internal.net Execute > bridge(sofia/gateway/${default_gateway}/70904) > 2009-03-16 13:35:39 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1000 at pbx-sip-3.usa.internal.net Expanded > String bridge(sofia/gateway/pbx-sip-4.usa.internal.net/70904) > 2009-03-16 13:35:39 [ERR] mod_sofia.c:2379 sofia_outgoing_channel() Invalid > Gateway > 2009-03-16 13:35:39 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() > Close Channel N/A [CS_NEW] > 2009-03-16 13:35:39 [ERR] switch_ivr_originate.c:1425 > switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [DEBUG] switch_ivr_originate.c:2014 > switch_ivr_originate() Originate Resulted in Error Cause: 28 > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [INFO] mod_dptools.c:1998 audio_bridge_function() > Originate Failed. Cause: INVALID_NUMBER_FORMAT > > > What else am I missing and not doing right? Thanks again for your help. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/1bdd6466/attachment.html From brian at freeswitch.org Mon Mar 16 10:53:32 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 12:53:32 -0500 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903161051k1c6609ecp1b0ec57a68810e8@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> <4ea6e8f20903161051k1c6609ecp1b0ec57a68810e8@mail.gmail.com> Message-ID: <44B6617A-ED52-44DF-82F1-B931BDD09F7F@freeswitch.org> First off since its not in the user directory anymore you'll have to unwrap the gateway from inside the user tags ;) /b On Mar 16, 2009, at 12:51 PM, Steven Ward wrote: > Sure thing. Here it is: > > > > > > > > > > > > > > > > > > > In vars.conf I supplied the variables' values: > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/250e0569/attachment.html From steve.d.ward at gmail.com Mon Mar 16 11:08:19 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 14:08:19 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <44B6617A-ED52-44DF-82F1-B931BDD09F7F@freeswitch.org> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> <4ea6e8f20903161051k1c6609ecp1b0ec57a68810e8@mail.gmail.com> <44B6617A-ED52-44DF-82F1-B931BDD09F7F@freeswitch.org> Message-ID: <4ea6e8f20903161108l2486ffbfl6f73acf6c298f7a6@mail.gmail.com> Heh heh. Guess it pays not to rush. :) Got it working now - without registering. But another thing - what if I want to set my two boxes up for registering? I see that I can set my register parameter to true, but how do I control the register string that's sent to the other box? 2009/3/16 Brian West > First off since its not in the user directory anymore you'll have to unwrap > the gateway from inside the user tags ;) > /b > > On Mar 16, 2009, at 12:51 PM, Steven Ward wrote: > > Sure thing. Here it is: > > > > > > > > value="$${default_provider_from_domain}"/> > > > > > > > > > > > In vars.conf I supplied the variables' values: > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/ac3f3aed/attachment-0001.html From kerrada2003 at yahoo.com Mon Mar 16 11:57:57 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 16 Mar 2009 11:57:57 -0700 (PDT) Subject: [Freeswitch-users] Outbound Codec Message-ID: <927033.2754.qm@web33705.mail.mud.yahoo.com> Hi, Is there a way to configure FS to offer a specific codec for B regardless of the codec chosen for? A leg? So, when FS invites B, it will offer only one codec specified by the administrator. Thanks, ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/4bcd12a7/attachment.html From brian at freeswitch.org Mon Mar 16 12:06:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 14:06:18 -0500 Subject: [Freeswitch-users] Outbound Codec In-Reply-To: <927033.2754.qm@web33705.mail.mud.yahoo.com> References: <927033.2754.qm@web33705.mail.mud.yahoo.com> Message-ID: <3CF232FA-7502-4A4C-B037-1DB65D410A61@freeswitch.org> {absolute_codec_string=PCMU}sofia/blah/blah at blah /b On Mar 16, 2009, at 1:57 PM, Ali Al-Rubaie wrote: > Hi, > > Is there a way to configure FS to offer a specific codec for B > regardless of the codec chosen for A leg? So, when FS invites B, it > will offer only one codec specified by the administrator. > > Thanks, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/c1027f65/attachment.html From sicfslist at gmail.com Mon Mar 16 12:07:01 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 16 Mar 2009 14:07:01 -0500 Subject: [Freeswitch-users] Outbound Codec In-Reply-To: <927033.2754.qm@web33705.mail.mud.yahoo.com> References: <927033.2754.qm@web33705.mail.mud.yahoo.com> Message-ID: <35b355e90903161207j4b136d68pbbc9d055ca3697fb@mail.gmail.com> Yes. You can set this in the sip profile settings ... and then when it's called in the bridge statement it will just work. For example you take a call on profile external (which allows multiple codecs) and then bridge it via profile internal (which only allows one codec). SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/48aefee2/attachment.html From ludovic.fouquet at bewan.com Mon Mar 16 10:55:25 2009 From: ludovic.fouquet at bewan.com (ludovic) Date: Mon, 16 Mar 2009 18:55:25 +0100 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? Message-ID: <49BE928D.3090509@bewan.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/3fbfae78/attachment.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: trace_fs.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/3fbfae78/attachment.txt From brian at freeswitch.org Mon Mar 16 13:08:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 15:08:18 -0500 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: <49BE928D.3090509@bewan.com> References: <49BE928D.3090509@bewan.com> Message-ID: This would be one thing to look at your DNS name isn't resolving correctly.. you might consider using dynamic DNS and you can then set the them to "host:myhost.dyndns.org" /b On Mar 16, 2009, at 12:55 PM, ludovic wrote: > 2009-03-16 18:29:42 [DEBUG] sofia.c:206 sofia_event_callback() event > [nua_r_invite] status [503][DNS Error] session: sofia/external/ > 0123456789 > 2009-03-16 18:29:42 [DEBUG] sofia.c:206 sofia_event_callback() event > [nua_i_state] status [503][DNS Error] session: sofia/external/ > 0123456789 From chris at fowler.cc Mon Mar 16 13:19:18 2009 From: chris at fowler.cc (Chris Fowler) Date: Mon, 16 Mar 2009 13:19:18 -0700 Subject: [Freeswitch-users] Possible memory / cpu leak Message-ID: <1237234758.32766.1305717113@webmail.messagingengine.com> Hi, I?ve been seeing an issue where FreeSWITCH?s CPU and memory utilization climb over time; a restart of FS clears up the problem. See graphs for the past week. http://cfowl.postinbox.com/fs.jpg Observed on the Release Candidate, and then upgraded to the current trunk a couple of times. Currently running version ?FreeSWITCH Version 1.0.trunk (12604)?. This is seen both when FS is being used (~200 calls/day, and over the weekend when ~5 calls/day). How can I best debug this? Thanks, Chris. From brian at freeswitch.org Mon Mar 16 14:37:28 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 16:37:28 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237234758.32766.1305717113@webmail.messagingengine.com> References: <1237234758.32766.1305717113@webmail.messagingengine.com> Message-ID: <81B45675-0B08-45FA-B0BB-A23C834A438B@freeswitch.org> Can you update to SVN trunk as of now? /b On Mar 16, 2009, at 3:19 PM, Chris Fowler wrote: > Hi, > > I?ve been seeing an issue where FreeSWITCH?s CPU and memory > utilization > climb over time; a restart of FS clears up the problem. > > See graphs for the past week. http://cfowl.postinbox.com/fs.jpg > > Observed on the Release Candidate, and then upgraded to the current > trunk a couple of times. Currently running version ?FreeSWITCH > Version > 1.0.trunk (12604)?. > > This is seen both when FS is being used (~200 calls/day, and over the > weekend when ~5 calls/day). > > How can I best debug this? > > Thanks, Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d16a6b7d/attachment-0001.html From brian at freeswitch.org Mon Mar 16 14:39:03 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 16:39:03 -0500 Subject: [Freeswitch-users] Don't forget about FreeSWITCH-biz mailing list. Message-ID: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz If you want to do business related posts... ads... discuss services and such that don't belong on the users or dev lists please join the freeswitch-dev list. Thanks, Brian West From jaybinks at gmail.com Mon Mar 16 15:27:29 2009 From: jaybinks at gmail.com (jay binks) Date: Tue, 17 Mar 2009 08:27:29 +1000 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237234758.32766.1305717113@webmail.messagingengine.com> References: <1237234758.32766.1305717113@webmail.messagingengine.com> Message-ID: what happens in your dialplan ? is is possible that you execute a script on each call, thats not being exited ? Jay On Tue, Mar 17, 2009 at 6:19 AM, Chris Fowler wrote: > Hi, > > I?ve been seeing an issue where FreeSWITCH?s CPU and memory utilization > climb over time; a restart of FS clears up the problem. > > See graphs for the past week. http://cfowl.postinbox.com/fs.jpg > > Observed on the Release Candidate, and then upgraded to the current > trunk a couple of times. Currently running version ?FreeSWITCH Version > 1.0.trunk (12604)?. > > This is seen both when FS is being used (~200 calls/day, and over the > weekend when ~5 calls/day). > > How can I best debug this? > > Thanks, Chris. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/b321575f/attachment.html From chris at fowler.cc Mon Mar 16 15:42:03 2009 From: chris at fowler.cc (Chris Fowler) Date: Mon, 16 Mar 2009 15:42:03 -0700 Subject: [Freeswitch-users] Possible memory / cpu leak Message-ID: <1237243323.1988.1305741229@webmail.messagingengine.com> >> Jay : what happens in your dialplan ? Nothing special; no external script execution just default pattern matching to route to extensions (per the stock config). >> Brian: Can you update to SVN trunk as of now? Yup, I will pull the trunk and report back in 24 hours. Chris. From brian at freeswitch.org Mon Mar 16 15:49:04 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 17:49:04 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237243323.1988.1305741229@webmail.messagingengine.com> References: <1237243323.1988.1305741229@webmail.messagingengine.com> Message-ID: Happen to use voicemail a lot? ivr.conf.xml? /b On Mar 16, 2009, at 5:42 PM, Chris Fowler wrote: >>> Jay : what happens in your dialplan ? > Nothing special; no external script execution just default pattern > matching to route to extensions (per the stock config). > >>> Brian: Can you update to SVN trunk as of now? > Yup, I will pull the trunk and report back in 24 hours. > > Chris. From anthony.minessale at gmail.com Mon Mar 16 15:51:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Mar 2009 17:51:50 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237243323.1988.1305741229@webmail.messagingengine.com> References: <1237243323.1988.1305741229@webmail.messagingengine.com> Message-ID: <191c3a030903161551i72789854m680978cd063701ba@mail.gmail.com> nothing special is a bit vague. clearly something you are doing makes a difference. Perhaps you can explain any custom extensions you have or what you are doing a little better? install valgrind and run it for a while valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg then send us vg.log On Mon, Mar 16, 2009 at 5:42 PM, Chris Fowler wrote: > >> Jay : what happens in your dialplan ? > Nothing special; no external script execution just default pattern > matching to route to extensions (per the stock config). > > >> Brian: Can you update to SVN trunk as of now? > Yup, I will pull the trunk and report back in 24 hours. > > Chris. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/607776d2/attachment.html From gservat at gmail.com Mon Mar 16 16:07:45 2009 From: gservat at gmail.com (Gonzalo Servat) Date: Mon, 16 Mar 2009 20:07:45 -0300 Subject: [Freeswitch-users] Don't forget about FreeSWITCH-biz mailing list. In-Reply-To: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> References: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> Message-ID: On Mon, Mar 16, 2009 at 6:39 PM, Brian West wrote: > http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz > > If you want to do business related posts... ads... discuss services > and such that don't belong on the users or dev lists please join the > freeswitch-dev list. > BUG! s/freeswitch-dev/freeswitch-biz ;-) - Gonzalo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/e6ce7b87/attachment.html From brian at freeswitch.org Mon Mar 16 16:39:10 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 18:39:10 -0500 Subject: [Freeswitch-users] Don't forget about FreeSWITCH-biz mailing list. In-Reply-To: References: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> Message-ID: <52853F7E-2F51-4916-9FE3-F72F76C515FF@freeswitch.org> Ah yes... my bad... Thats what I get for doing two more things at once... Had a thread in my brain blocking! :P /b On Mar 16, 2009, at 6:07 PM, Gonzalo Servat wrote: > On Mon, Mar 16, 2009 at 6:39 PM, Brian West > wrote: > http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz > > If you want to do business related posts... ads... discuss services > and such that don't belong on the users or dev lists please join the > freeswitch-dev list. > > BUG! s/freeswitch-dev/freeswitch-biz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/43b33f17/attachment.html From dujinfang at gmail.com Mon Mar 16 19:01:09 2009 From: dujinfang at gmail.com (seven) Date: Tue, 17 Mar 2009 10:01:09 +0800 Subject: [Freeswitch-users] FS DB In-Reply-To: <595500.67715.qm@web33706.mail.mud.yahoo.com> References: <595500.67715.qm@web33706.mail.mud.yahoo.com> Message-ID: <7BA5277B-9A6A-42C6-9388-3898E619C51A@gmail.com> In default configuration, all DB is located in FS/db/. It's sqlite DB, so if you have sqlite installed # apt-get install sqlite3 $ cd /usr/local/freeswitch/db $ sqlite3 core.db SQLite version 3.4.2 Enter ".help" for instructions sqlite> .help .bail ON|OFF Stop after hitting an error. Default OFF .databases List names and files of attached databases .dump ?TABLE? ... Dump the database in an SQL text format .echo ON|OFF Turn command echo on or off .exit Exit this program .explain ON|OFF Turn output mode suitable for EXPLAIN on or off. .header(s) ON|OFF Turn display of headers on or off .help Show this message .import FILE TABLE Import data from FILE into TABLE .indices TABLE Show names of all indices on TABLE .load FILE ?ENTRY? Load an extension library .mode MODE ?TABLE? Set output mode where MODE is one of: csv Comma-separated values column Left-aligned columns. (See .width) html HTML code insert SQL insert statements for TABLE line One value per line list Values delimited by .separator string tabs Tab-separated values tcl TCL list elements .nullvalue STRING Print STRING in place of NULL values .output FILENAME Send output to FILENAME .output stdout Send output to the screen .prompt MAIN CONTINUE Replace the standard prompts .quit Exit this program .read FILENAME Execute SQL in FILENAME .schema ?TABLE? Show the CREATE statements .separator STRING Change separator used by output mode and .import .show Show the current values for various settings .tables ?PATTERN? List names of tables matching a LIKE pattern .timeout MS Try opening locked tables for MS milliseconds .width NUM NUM ... Set column widths for "column" mode sqlite> .tables aliases calls channels complete interfaces tasks sqlite> .schema channels CREATE TABLE channels ( uuid VARCHAR(255), created VARCHAR(255), created_epoch INTEGER, name VARCHAR(255), state VARCHAR(255), cid_name VARCHAR(255), cid_num VARCHAR(255), ip_addr VARCHAR(255), dest VARCHAR(255), application VARCHAR(255), application_data VARCHAR(255), dialplan VARCHAR(255), context VARCHAR(255), read_codec VARCHAR(255), read_rate VARCHAR(255), write_codec VARCHAR(255), write_rate VARCHAR(255) ); CREATE INDEX uuindex on channels (uuid); sqlite> select * from channels ...> ; sqlite> On Mar 17, 2009, at 12:46 AM, Ali Al-Rubaie wrote: > > Actually, I'm trying to explore what data is stored in the DB and > how can we retrieve it, if it is posible. You may write some scripts > to retrieve the data and use it in other supporting application. > > > Message: 7 > Date: Mon, 16 Mar 2009 12:24:34 -0400 > From: Michael Jerris > Subject: Re: [Freeswitch-users] FS Database > To: freeswitch-users at lists.freeswitch.org > Message-ID: <1BF9A176-1CE8-4D53-894B-E456DA8897A0 at jerris.com> > Content-Type: text/plain; charset="us-ascii" > > Which data? > > On Mar 16, 2009, at 11:58 AM, Ali Al-Rubaie wrote: > > > Hi, > > > > Is it possible to access FS DB to retrieve data? Where can i find > > details about that? > -------------- next part > -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d0d0a1b0/attachment.html > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 33, Issue 67 > ************************************************ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/a7cc7e13/attachment-0001.html From dujinfang at gmail.com Mon Mar 16 19:06:23 2009 From: dujinfang at gmail.com (seven) Date: Tue, 17 Mar 2009 10:06:23 +0800 Subject: [Freeswitch-users] Don't forget about FreeSWITCH-biz mailing list. In-Reply-To: References: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> Message-ID: <118441D4-BC68-4969-8BA8-AB2076E9F946@gmail.com> It's time to open a jira :D On Mar 17, 2009, at 7:07 AM, Gonzalo Servat wrote: > On Mon, Mar 16, 2009 at 6:39 PM, Brian West > wrote: > http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz > > If you want to do business related posts... ads... discuss services > and such that don't belong on the users or dev lists please join the > freeswitch-dev list. > > BUG! s/freeswitch-dev/freeswitch-biz > > ;-) > > - Gonzalo > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/c250d52c/attachment.html From chris at fowler.cc Mon Mar 16 21:01:10 2009 From: chris at fowler.cc (Chris Fowler) Date: Mon, 16 Mar 2009 21:01:10 -0700 Subject: [Freeswitch-users] Possible memory / cpu leak Message-ID: <1237262470.17561.1305780929@webmail.messagingengine.com> >> Brian: Can you update to SVN trunk as of now? I updated - version reports: FreeSWITCH Version 1.0.trunk (12631) Only difference I note with this build is that upon "shutdown" FS now SegFaults. The mem/cpu usage continues to slowly climb. 2009-03-16 20:59:32 [CONSOLE] switch_loadable_module.c:1237 do_shutdown() Stopping: mod_spidermonkey Segmentation fault (core dumped) >> Anthony - nothing special is a bit vague. I modified the dial plan to accept extension in the 1000-1029 range Added DialByLast name using directory.lua (from the wiki) Modified the ivr config to play company specific greetings Voicemail is used a few times per day freeswitch at ip-10-250-155-18> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= internal profile sip:mod_sofia at xxx.xxx.xxx.xx:5060 RUNNING (0) external profile sip:mod_sofia at xxx.xxx.xxx.xx:5080 RUNNING (0) sip.flowroute.com gateway sip:xxxxxx at sip.flowroute.com REGED inphonex gateway sip:xxxxx at sip.inphonex.com REGED callwithus-did-xxxxxxxxxx gateway sip:xxxxxxx at east.callwithus.com REGED callwithus-did-xxxxxxxxxx gateway sip:xxxxxxx at east.callwithus.com REGED callwithus-did-xxxxxxxxxx gateway sip:xxxxxxx at east.callwithus.com REGED default alias internal ALIASED nat alias external ALIASED xxxxx.bbbb.aaa alias internal ALIASED outbound alias external ALIASED ================================================================================================= >> Anthony - valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg Nothing got logged, here's the output. Did I invoke valgrind incorrectly? ==32545== Memcheck, a memory error detector. ==32545== Copyright (C) 2002-2006, and GNU GPL'd, by Julian Seward et al. ==32545== Using LibVEX rev 1658, a library for dynamic binary translation. ==32545== Copyright (C) 2004-2006, and GNU GPL'd, by OpenWorks LLP. ==32545== Using valgrind-3.2.1, a dynamic binary instrumentation framework. ==32545== Copyright (C) 2000-2006, and GNU GPL'd, by Julian Seward et al. ==32545== For more details, rerun with: -v ==32545== ==32545== My PID = 32545, parent PID = 32511. Prog and args are: ==32545== /usr/local/freeswitch/bin/freeswitch ==32545== -vg ==32545== From brian at freeswitch.org Mon Mar 16 21:09:49 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 23:09:49 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237262470.17561.1305780929@webmail.messagingengine.com> References: <1237262470.17561.1305780929@webmail.messagingengine.com> Message-ID: Tip: ulimit -s 240, or freeswitch fork and execv out from under valgrind. /b On Mar 16, 2009, at 11:01 PM, Chris Fowler wrote: > > >>> Anthony - valgrind --tool=memcheck --log-file=vg.log --leak- >>> check=full --leak-resolution=high --show-reachable=yes /path/to/ >>> freeswitch -vg > > > Nothing got logged, here's the output. Did I invoke valgrind > incorrectly? > > ==32545== Memcheck, a memory error detector. > ==32545== Copyright (C) 2002-2006, and GNU GPL'd, by Julian Seward et > al. > ==32545== Using LibVEX rev 1658, a library for dynamic binary > translation. > ==32545== Copyright (C) 2004-2006, and GNU GPL'd, by OpenWorks LLP. > ==32545== Using valgrind-3.2.1, a dynamic binary instrumentation > framework. > ==32545== Copyright (C) 2000-2006, and GNU GPL'd, by Julian Seward et > al. > ==32545== For more details, rerun with: -v > ==32545== > ==32545== My PID = 32545, parent PID = 32511. Prog and args are: > ==32545== /usr/local/freeswitch/bin/freeswitch > ==32545== -vg > ==32545== From tntknight at gmail.com Mon Mar 16 21:55:35 2009 From: tntknight at gmail.com (Anthony Knight) Date: Tue, 17 Mar 2009 00:55:35 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards Message-ID: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> I'm thinking about doing a project that would use FreeSWITCH as an IVR, with callers being routed in by both ISDN PRI, and also SIP trunks, with occasional bridge calls between callers. I'm wondering in what use cases hardware echo cancellation on the PRI cards is needed. And does hardware echo cancellation work with OpenZap/FreeSWITCH? It looks like all the major cards (Sangoma, Digium, etc..) use Octasic Echo cancellation add-on cards. Is there any difference between brands? Any recommendations on PRI boards and whether I need to pay for echo cancellation are appreciated Thanks. Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/783854d7/attachment.html From krice at freeswitch.org Mon Mar 16 22:35:53 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 17 Mar 2009 00:35:53 -0500 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> Message-ID: The cards that feature Hardware Echo Can?s work on hardware/driver level and are supported... From: Anthony Knight Reply-To: Date: Tue, 17 Mar 2009 00:55:35 -0400 To: Subject: [Freeswitch-users] echo cancellation on PRI cards I'm thinking about doing a project that would use?FreeSWITCH?as an IVR, with callers being routed in by both ISDN PRI, and also SIP trunks, with occasional bridge calls between callers. I'm wondering in what use cases hardware echo cancellation on the PRI cards is needed. ?And does hardware echo cancellation work with OpenZap/FreeSWITCH?? It looks like all the major cards (Sangoma, Digium, etc..) use Octasic Echo cancellation add-on cards. ?Is there any difference between brands? Any recommendations on PRI boards and whether I need to pay for echo cancellation are appreciated Thanks. Tony _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/5f7e30eb/attachment.html From wasim at convergence.pk Mon Mar 16 22:48:20 2009 From: wasim at convergence.pk (Wasim Baig) Date: Tue, 17 Mar 2009 10:48:20 +0500 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> Message-ID: 2009/3/17 Anthony Knight I'm thinking about doing a project that would use FreeSWITCH as an IVR, with > callers being routed in by both ISDN PRI, and also SIP trunks, with > occasional bridge calls between callers. > > I'm wondering in what use cases hardware echo cancellation on the PRI cards > is needed. > When there is Echo being generated from the far end, usually in a bridged call. If you application is just an IVR, with no far end connectivity, then you shouldn't need an echo can. If you are bridging calls, then at some point you may need it, depending on what else is in the loop. > And does hardware echo cancellation work with OpenZap/FreeSWITCH? > Yes, it really has nothing to do with the software then, its handled by the card and its hardware driver. In Sangoma's case, by Wanpipe. > It looks like all the major cards (Sangoma, Digium, etc..) use Octasic Echo > cancellation add-on cards. Is there any difference between brands? > Sangoma has 1024 tap Octasic Echo Cans. Very nice they are indeed. > Any recommendations on PRI boards and whether I need to pay for echo > cancellation are appreciated > Unashamedly, Sangoma's. 100% of the cases where our customers have used Sangoma A10Xd vs A10X, they've been much happier with the quality on the line. Its a tad bit more $, but well worth it (especially in places with bad copper). -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/96550390/attachment.html From kawarod at laposte.net Tue Mar 17 00:06:29 2009 From: kawarod at laposte.net (rod) Date: Tue, 17 Mar 2009 11:06:29 +0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA844C.3010409@freeswitch.org> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> <49BA6608.70208@gmail.com> <49BA844C.3010409@freeswitch.org> Message-ID: <49BF4BF5.4080608@laposte.net> Hi, not too hard :p but it's just a bad habit when I write in my native language (french). I guess that this spelling is not too common for english speaker. I'll do my best next time to write it correctly. @tamas you are right, we could use limit_hash the same way as limit when not specifying the /rate @Mathieu did you suggest limit_hash is more scalable than limit? But I don't understand why limit_hash is not suitable for data replication (DB lookup for limit and memory for limit_hash??), even if I don't know how to do it with limit. regards. Raymond Chandler wrote: > Tamas wrote: > >> My guess is: pbm = problem :) >> >> > sure, but is it really that hard to spell all the way out? > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From kawarod at laposte.net Tue Mar 17 00:11:26 2009 From: kawarod at laposte.net (rod) Date: Tue, 17 Mar 2009 11:11:26 +0400 Subject: [Freeswitch-users] sip redirect contact variable no more available in SVN 12638 Message-ID: <49BF4D1E.9050108@laposte.net> Hi, running SVN r12638, I don't have access anymore to these 2 variables after a SIP 302 message, using info application: variable_sip_redirect_contact_user_0 variable_sip_redirect_contact_host_0 It was okay with SVN r12611. regards, rod From codecomplete at free.fr Tue Mar 17 02:02:29 2009 From: codecomplete at free.fr (Gilles) Date: Tue, 17 Mar 2009 10:02:29 +0100 Subject: [Freeswitch-users] Windows-compatible FXO PCI card? Message-ID: <7.0.1.0.2.20090317100224.02475630@free.fr> Hello For SOHO users, getting a second, Linux-based computer just to run a small voice server is overkill, so I'm thinking of selling an application based on the Windows version of Freeswitch. Instead of the Sangoma USB connector, I'd really prefer to sell them a PCI card, because it's less messy, and there's less chance of them disconnecting the hardware. I don't know of FXO PCI cards to connect an XP/Vista host to a phone line. Does someone know of such a thing? Thank you. From wasim at convergence.pk Tue Mar 17 02:21:42 2009 From: wasim at convergence.pk (Wasim Baig) Date: Tue, 17 Mar 2009 14:21:42 +0500 Subject: [Freeswitch-users] Windows-compatible FXO PCI card? In-Reply-To: <7.0.1.0.2.20090317100224.02475630@free.fr> References: <7.0.1.0.2.20090317100224.02475630@free.fr> Message-ID: On Tue, Mar 17, 2009 at 2:02 PM, Gilles wrote: I don't know of FXO PCI cards to connect an XP/Vista host to a phone > line. Does someone know of such a thing? Sangoma makes a low cost 4FXO, 1FXS. http://sangoma.com/products_and_solutions/hardware/analog_telephony/b600.html -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/273b5c79/attachment.html From thomas.mangin at exa-networks.co.uk Tue Mar 17 02:27:13 2009 From: thomas.mangin at exa-networks.co.uk (Thomas Mangin) Date: Tue, 17 Mar 2009 09:27:13 +0000 Subject: [Freeswitch-users] Freeswitch and Kamailio (OpenSer) Integration In-Reply-To: <439e75680903060051y5021b292l76286dfe927d0337@mail.gmail.com> References: <439e75680903060051y5021b292l76286dfe927d0337@mail.gmail.com> Message-ID: <17216626-2A95-4BB1-8AD6-E7CD2C2B402C@exa-networks.co.uk> Yes it is possible but there is no documentation on how to do it. You will have to learn SIP and understand what you are doing. Forwarding the call to FS for nat may cause issue as FS will then not have direct connection to the phone and may not be able to always detect it is behind NAT. Have a look at SIP PATH Extension as it is what you need to force traffic back to KAM from FS. Regards, Thomas On 6 Mar 2009, at 08:51, Ramu wrote: > Hi All, > > I would like to setup freswitch and kamailio as follows: > > Kamailio acts as Proxy and Registrator > Freeswitch acts as a SBC and MediaServer (voicemail) > > Users will be reigstered to Kamailio > Kamailio forwards calls to FS to NAT > FS sends back INVITE to Kamailio > Kamailio will dial-out user. > > Bob calls Alice > Bob ==INVITE ==> Kamailio ==INVITE==> FS ==INVITE==> Kamailio > ==INVITE ==> Alice > > How can I achieve this scenario? Can you please direct me to any > documentation which is available? > > Thanks, > Ramu > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/8b68d0b7/attachment.html From Mark.Tabron at rnid-typetalk.org.uk Tue Mar 17 02:31:01 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Tue, 17 Mar 2009 09:31:01 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though. I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it! Mark. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 16 March 2009 17:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Any chance you can give one of us access to this system? Best thing to do would be to join #openzap on irc.freenode.net. -MC (IRC: mercutioviz) On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron wrote: > Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room. > > So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be. > > Any ideas? Pastebin debug output is in my reply below. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron > Sent: 13 March 2009 14:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). > > Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! > > http://pastebin.freeswitch.org/7751 > > Will setup an IRC client and see if I can log onto the channel. > > Thanks again! > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 12 March 2009 16:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > >> My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so >> apologies if some of my terminology isn't quite correct. > > Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > >> >> >> >> Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've >> hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the >> card is green and wanrouter is installed and up in TDM_API mode, with the >> connection status showing as connected. ?Configured Openzap for 9 b and 1 d >> channel as described in Freeswitch Wiki. Then created a diaplan to fire off >> any calls preceded by 9 to the next available openzap channel. > > Looks good so far... > >> The problem I have is when I initiate an external call (using 9xxxxxxx) from >> an extension I can see Freeswitch allocating the call to the next available >> channel but then the just sits there and times out after 1 minute. With the >> cause stated as ORIGINATOR_CANCEL (guessing this is the time out) > > okay, some debugging info will be useful. Please read this wiki page first: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It has lots of useful information for how to gather log information, > how to use the pastebin, etc. > > Specifically for this issue you'll need to use the pastebin because > there will be so much information. Here are some pointers: > > To see what's happening with openzap you'll need to use the "oz list" > and "oz dump 1" at the command line (CLI). You'll also need to turn on > debugging so that PRI messages show up. You'll need to capture the > output on the CLI and put it into the pastebin. > (http://pastebin.freeswitch.org). > > Welcome to the wonderful world of telephony debugging! > -MC > > P.S. - We have a few IRC channels where you can join to get more > real-time support: > #freeswitch and #openzap on irc.freenode.net. (More details are in the > wiki page I mentioned above.) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > -------------------------------------------------------------------------------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > -------------------------------------------------------------------------------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mrene_lists at avgs.ca Tue Mar 17 03:10:57 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Mar 2009 06:10:57 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BF4BF5.4080608@laposte.net> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> <49BA6608.70208@gmail.com> <49BA844C.3010409@freeswitch.org> <49BF4BF5.4080608@laposte.net> Message-ID: <85B32876-7E49-4DDF-B92F-353DA9599DE9@avgs.ca> limit_hash uses a faster data structure then limit but works the same way for tne end-user. viens sur IRC si t'as des questions en francais =) Math On 17-Mar-09, at 3:06 AM, rod wrote: > Hi, > > not too hard :p > but it's just a bad habit when I write in my native language > (french). I > guess that this spelling is not too common for english speaker. > > I'll do my best next time to write it correctly. > > @tamas > you are right, we could use limit_hash the same way as limit when not > specifying the /rate > > @Mathieu > did you suggest limit_hash is more scalable than limit? But I don't > understand why limit_hash is not suitable for data replication (DB > lookup for limit and memory for limit_hash??), even if I don't know > how > to do it with limit. > > regards. > > Raymond Chandler wrote: >> Tamas wrote: >> >>> My guess is: pbm = problem :) >>> >>> >> sure, but is it really that hard to spell all the way out? >> >> -Ray >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ludovic.fouquet at bewan.com Tue Mar 17 04:22:13 2009 From: ludovic.fouquet at bewan.com (ludovic) Date: Tue, 17 Mar 2009 12:22:13 +0100 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: References: <49BE928D.3090509@bewan.com> Message-ID: <49BF87E5.5090809@bewan.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/0a4d5052/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: bewan100.jpg Type: image/jpeg Size: 3963 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/0a4d5052/attachment-0001.jpg From Mark.Tabron at rnid-typetalk.org.uk Tue Mar 17 04:24:31 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Tue, 17 Mar 2009 11:24:31 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 17 March 2009 09:31 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though. I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it! Mark. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 16 March 2009 17:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Any chance you can give one of us access to this system? Best thing to do would be to join #openzap on irc.freenode.net. -MC (IRC: mercutioviz) On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron wrote: > Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room. > > So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be. > > Any ideas? Pastebin debug output is in my reply below. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron > Sent: 13 March 2009 14:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). > > Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! > > http://pastebin.freeswitch.org/7751 > > Will setup an IRC client and see if I can log onto the channel. > > Thanks again! > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 12 March 2009 16:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > >> My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so >> apologies if some of my terminology isn't quite correct. > > Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > >> >> >> >> Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've >> hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the >> card is green and wanrouter is installed and up in TDM_API mode, with the >> connection status showing as connected. ?Configured Openzap for 9 b and 1 d >> channel as described in Freeswitch Wiki. Then created a diaplan to fire off >> any calls preceded by 9 to the next available openzap channel. > > Looks good so far... > >> The problem I have is when I initiate an external call (using 9xxxxxxx) from >> an extension I can see Freeswitch allocating the call to the next available >> channel but then the just sits there and times out after 1 minute. With the >> cause stated as ORIGINATOR_CANCEL (guessing this is the time out) > > okay, some debugging info will be useful. Please read this wiki page first: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It has lots of useful information for how to gather log information, > how to use the pastebin, etc. > > Specifically for this issue you'll need to use the pastebin because > there will be so much information. Here are some pointers: > > To see what's happening with openzap you'll need to use the "oz list" > and "oz dump 1" at the command line (CLI). You'll also need to turn on > debugging so that PRI messages show up. You'll need to capture the > output on the CLI and put it into the pastebin. > (http://pastebin.freeswitch.org). > > Welcome to the wonderful world of telephony debugging! > -MC > > P.S. - We have a few IRC channels where you can join to get more > real-time support: > #freeswitch and #openzap on irc.freenode.net. (More details are in the > wiki page I mentioned above.) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > -------------------------------------------------------------------------------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > -------------------------------------------------------------------------------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From Prometheus001 at gmx.net Tue Mar 17 04:27:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 17 Mar 2009 12:27:25 +0100 Subject: [Freeswitch-users] Problem with shortened local extensions In-Reply-To: References: <87f2f3b90903160906q693370c8s6b68eafd528f6401@mail.gmail.com> Message-ID: <49BF891D.1090808@gmx.net> I also had problems with not reaching local extensions some time. I solved it by adding: to /usr/local/freeswitch/conf/directory/default.xml Best regards Peter fs at xenpad.eu schrieb: > Hi, > > On Mon, 16 Mar 2009, Michael Collins wrote: > >>> I have a (probably dumb) question that I just spent over 5 hours on: >>> I have a 1.0.2 version running with 6 extensions, 2 SIP trunks and >>> gtalk OK. >> >> Ouch! Any way you could update? We are on the verge of releasing >> 1.0.4; 1.0.2 is OLD. :) > > Then I'll skip 1.0.3 and wait for 1.0.4 ;) > >> That will help you do stuff like this: >> turn on debugging (press F8 at the CLI) then make a test call, capture >> output, put it into a pastebin. > OK -- this is a live system; I worked on it this weekend. > I'll try again with the short extension and collect all data. > Should 1.0.4 be out in the meantime, I'll upgrade before I try. > > Cheers, > Laurent > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveu at coppice.org Tue Mar 17 05:14:34 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 17 Mar 2009 20:14:34 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> Message-ID: <49BF942A.3030305@coppice.org> Wasim Baig wrote: > 2009/3/17 Anthony Knight > > > I'm thinking about doing a project that would use FreeSWITCH as an > IVR, with callers being routed in by both ISDN PRI, and also SIP > trunks, with occasional bridge calls between callers. > > I'm wondering in what use cases hardware echo cancellation on the > PRI cards is needed. > > > When there is Echo being generated from the far end, usually in a > bridged call. If you application is just an IVR, with no far end > connectivity, then you shouldn't need an echo can. If you are bridging > calls, then at some point you may need it, depending on what else is > in the loop. This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it they give very poor reliability detecting DTMF while the prompts are playing. If the system uses voice recognition, its reliability will be even worse. > And does hardware echo cancellation work with OpenZap/FreeSWITCH? > > > Yes, it really has nothing to do with the software then, its handled > by the card and its hardware driver. In Sangoma's case, by Wanpipe. > > > It looks like all the major cards (Sangoma, Digium, etc..) use > Octasic Echo cancellation add-on cards. Is there any difference > between brands? > > > Sangoma has 1024 tap Octasic Echo Cans. Very nice they are indeed. > > > Any recommendations on PRI boards and whether I need to pay for > echo cancellation are appreciated > > > Unashamedly, Sangoma's. 100% of the cases where our customers have > used Sangoma A10Xd vs A10X, they've been much happier with the quality > on the line. Its a tad bit more $, but well worth it (especially in > places with bad copper). If you use Sangoma make sure everything is up to date. People have had a lot of DTMF detection trouble with some revisions of the driver, or on board firmware, or possibly both. Clearly DTMF trouble would be pretty bad for an IVR. I didn't manage to trace which were the offending versions, but the current stuff is apparently OK. Steve From brian at freeswitch.org Tue Mar 17 06:53:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 08:53:07 -0500 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: <49BF87E5.5090809@bewan.com> References: <49BE928D.3090509@bewan.com> <49BF87E5.5090809@bewan.com> Message-ID: What I provided you was an example. I don't think you understood what I was talking about. In the settings for ext-sip-ip and ext-rtp-ip you'll have to use something like "host:yourdyndnshostname.blah.tld" Then set the sip-ip and rtp-ip to what ever is auto detected. /b On Mar 17, 2009, at 6:22 AM, ludovic wrote: > Thanks. > It seems that it comes from my sip provider. > when using my_host as my hostname, reg fails > when using my_host.com as my hostname, reg succeeds (my_host.com > does not exist as a domain internet) > when using ip address, reg succeeds. > > Tested with version 1.0.3 > > Is it a way to force the IP address to be used in SIP header instead > of hostname ? > > Thanks > > Ludovic From kzimnicki at gcdf.pl Tue Mar 17 02:17:13 2009 From: kzimnicki at gcdf.pl (Krzysztof Zimnicki) Date: Tue, 17 Mar 2009 10:17:13 +0100 Subject: [Freeswitch-users] [OpenZap] problem with TE220P Message-ID: <49BF6A99.6070308@gcdf.pl> Hi, I have problem with Digium TE220P. Everything works, i can call & talk, but everytime i have CRIT message: 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 When FS start show me ERR message: 2009-03-14 17:44:06 [ERR] Span:0 Q.921() Received UA frame in invalid state This is my config: cat zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span = 2,0,0,ccs,hdb3,crc4 bchan = 32-46,48-62 dchan = 47 loadzone = ru defaultzone=ru cat openzap.conf [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 cat openzap.conf.xml and FS start LOGS: 2009-03-14 17:44:06 [NOTICE] zap_io.c:2612 zap_global_init() Modules configured: 1 2009-03-14 17:44:06 [NOTICE] ozmod_zt.c:922 zt_init() Using Zaptel control device 2009-03-14 17:44:06 [INFO] zap_io.c:2433 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_zt.so [zt] 2009-03-14 17:44:06 [INFO] zap_io.c:2233 load_config() auto-loaded 'zt' 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:38 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 2 as OpenZAP device 1:2 fd:39 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 3 as OpenZAP device 1:3 fd:40 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 4 as OpenZAP device 1:4 fd:41 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 5 as OpenZAP device 1:5 fd:42 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 6 as OpenZAP device 1:6 fd:43 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 7 as OpenZAP device 1:7 fd:44 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 8 as OpenZAP device 1:8 fd:45 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 9 as OpenZAP device 1:9 fd:46 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 10 as OpenZAP device 1:10 fd:47 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 11 as OpenZAP device 1:11 fd:48 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 12 as OpenZAP device 1:12 fd:49 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 13 as OpenZAP device 1:13 fd:50 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 14 as OpenZAP device 1:14 fd:51 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 15 as OpenZAP device 1:15 fd:52 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 16 as OpenZAP device 1:16 fd:53 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 17 as OpenZAP device 1:17 fd:54 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 18 as OpenZAP device 1:18 fd:55 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 19 as OpenZAP device 1:19 fd:56 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 20 as OpenZAP device 1:20 fd:57 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 21 as OpenZAP device 1:21 fd:58 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 22 as OpenZAP device 1:22 fd:59 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 23 as OpenZAP device 1:23 fd:60 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 24 as OpenZAP device 1:24 fd:61 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 25 as OpenZAP device 1:25 fd:62 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 26 as OpenZAP device 1:26 fd:63 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 27 as OpenZAP device 1:27 fd:64 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 28 as OpenZAP device 1:28 fd:65 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 29 as OpenZAP device 1:29 fd:66 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 30 as OpenZAP device 1:30 fd:67 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 31 as OpenZAP device 1:31 fd:68 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 32 as OpenZAP device 2:1 fd:69 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 33 as OpenZAP device 2:2 fd:70 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 34 as OpenZAP device 2:3 fd:71 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 35 as OpenZAP device 2:4 fd:72 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 36 as OpenZAP device 2:5 fd:73 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 37 as OpenZAP device 2:6 fd:74 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 38 as OpenZAP device 2:7 fd:75 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 39 as OpenZAP device 2:8 fd:76 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 40 as OpenZAP device 2:9 fd:77 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 41 as OpenZAP device 2:10 fd:78 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 42 as OpenZAP device 2:11 fd:79 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 43 as OpenZAP device 2:12 fd:80 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 44 as OpenZAP device 2:13 fd:81 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 45 as OpenZAP device 2:14 fd:82 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 46 as OpenZAP device 2:15 fd:83 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 47 as OpenZAP device 2:16 fd:84 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 48 as OpenZAP device 2:17 fd:85 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 49 as OpenZAP device 2:18 fd:86 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 50 as OpenZAP device 2:19 fd:87 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 51 as OpenZAP device 2:20 fd:88 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 52 as OpenZAP device 2:21 fd:89 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 53 as OpenZAP device 2:22 fd:90 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 54 as OpenZAP device 2:23 fd:91 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 55 as OpenZAP device 2:24 fd:92 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 56 as OpenZAP device 2:25 fd:93 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 57 as OpenZAP device 2:26 fd:94 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 58 as OpenZAP device 2:27 fd:95 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 59 as OpenZAP device 2:28 fd:96 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 60 as OpenZAP device 2:29 fd:97 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 61 as OpenZAP device 2:30 fd:98 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 62 as OpenZAP device 2:31 fd:99 2009-03-14 17:44:06 [INFO] zap_io.c:2356 load_config() Configured 62 channel(s) 2009-03-14 17:44:06 [INFO] zap_io.c:2450 zap_load_module() Loading SIG from /usr/local/freeswitch/mod/ozmod_isdn.so 2009-03-14 17:44:06 [INFO] zap_io.c:2566 zap_configure_span() auto-loaded 'isdn' 2009-03-14 17:44:06 [CONSOLE] switch_loadable_module.c:863 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:142 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:240 switch_loadable_module_process() Adding Application 'disable_ec' 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:260 switch_loadable_module_process() Adding API Function 'oz' 2009-03-14 17:44:06 [ERR] Span:0 Q.921() Received UA frame in invalid state When i dial from onference: freeswitch at voipgw> conference test at moh dial openzap/1/3/81048663000000 2009-03-14 17:50:26 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel OpenZAP/1:4/81048663000000 [d6137f1a-10af-11de-8496-c9e9c714d68b] 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 2009-03-14 17:50:30 [NOTICE] mod_openzap.c:1500 on_clear_channel_signal() Ring-Ready OpenZAP/1:4/81048663000000! 2009-03-14 17:50:34 [NOTICE] mod_openzap.c:1491 on_clear_channel_signal() Pre-Answer OpenZAP/1:4/81048663000000! 2009-03-14 17:50:35 [NOTICE] mod_openzap.c:1480 on_clear_channel_signal() Channel [OpenZAP/1:4/81048663000000] has been answered API CALL [conference(test at moh dial openzap/1/4/81048663000000)] output: Call Requested: result: [SUCCESS] 2009-03-14 17:44:06 [CONSOLE] switch_loadable_module.c:863 switch_loadable_module_load_file() Successfully Loaded [mod_commands] freeswitch at voipgw> oz list API CALL [oz(list)] output: +OK span: 1 (span1) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none freeswitch at voipgw> oz dump 1 API CALL [oz(dump 1)] output: +OK span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE span_id: 1 chan_id: 2 physical_span_id: 1 physical_chan_id: 2 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE span_id: 1 chan_id: 3 physical_span_id: 1 physical_chan_id: 3 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE [...] any idea how i can fix this error ? Krzysztof Zimnicki From pereyra.roberto at gmail.com Tue Mar 17 05:26:29 2009 From: pereyra.roberto at gmail.com (Roberto Pereyra) Date: Tue, 17 Mar 2009 09:26:29 -0300 Subject: [Freeswitch-users] enable anonymous incomming calls In-Reply-To: References: Message-ID: Hi all I'm freswitch newbie ?and have a simple question. How can enable anonymous inbound calls ? I would like to use freeswitch to accept incomming calls from sipbroker DIDs Any hint ? Thank in advance for all freeswitch team !! roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar The best dedicated servers - LiquidWeb http://www.liquidweb.com/?RID=contenid From matt at hellohunter.com Tue Mar 17 05:58:24 2009 From: matt at hellohunter.com (Matt Hunter) Date: Tue, 17 Mar 2009 19:58:24 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup Message-ID: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the "fifo in" execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/7d95ee3c/attachment.html From mattdfong at gmail.com Tue Mar 17 06:37:33 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 17 Mar 2009 20:37:33 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> Message-ID: <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the "fifo in" execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/50e92e38/attachment.html From dujinfang at gmail.com Tue Mar 17 07:50:02 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 17 Mar 2009 22:50:02 +0800 Subject: [Freeswitch-users] enable anonymous incomming calls In-Reply-To: References: Message-ID: <58C283C2-52C3-48E0-B124-999CB01842FC@gmail.com> at default config, in conf/sip_profiles/external.xml where $${external_sip_port} is a variable you can find in conf/ vars.xml, normally it's 5080, make sure sipbroker route calls to that port, and then you can make a dialplan in conf/dialplan/public.xml turn on verbose log in console by fs> console loglevel debug you can see the process when FS hit the dialplan On Mar 17, 2009, at 8:26 PM, Roberto Pereyra wrote: > Hi all > > I'm freswitch newbie and have a simple question. > > How can enable anonymous inbound calls ? I would like to use > freeswitch to accept incomming calls from sipbroker DIDs > > Any hint ? > > Thank in advance for all freeswitch team !! > > roberto > > -- > Ing. Roberto Pereyra > ContenidosOnline > http://www.contenidosonline.com.ar > > The best dedicated servers - LiquidWeb > http://www.liquidweb.com/?RID=contenid > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave at 3c.co.uk Tue Mar 17 08:04:29 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Mar 2009 09:04:29 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BF942A.3030305@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> Message-ID: <49BFBBFD.1050308@3c.co.uk> Steve Underwood wrote: >> When there is Echo being generated from the far end, usually in a >> bridged call. If you application is just an IVR, with no far end >> connectivity, then you shouldn't need an echo can. If you are bridging >> calls, then at some point you may need it, depending on what else is >> in the loop. >> > This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it > they give very poor reliability detecting DTMF while the prompts are > playing. If the system uses voice recognition, its reliability will be > even worse. > With respect, this is at best half true. DTMF detection has always worked just fine without echo cancellation - the Dialogic, Aculab and Rhetorex cards which I used in the late 1990s managed it perfectly well; if the DTMF detection code in * and FS can't, then maybe that's something for its author to look at ;-) ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the same hardware above back in the day. I'd be interested to see results of testing an ASR engine in with echo; unfortunately, most vendors appear to prohibit the publication of test results in their licensing. --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/061cb3cb/attachment.html From pereyra.roberto at gmail.com Tue Mar 17 08:16:35 2009 From: pereyra.roberto at gmail.com (Roberto Pereyra) Date: Tue, 17 Mar 2009 12:16:35 -0300 Subject: [Freeswitch-users] enable anonymous incomming calls In-Reply-To: <58C283C2-52C3-48E0-B124-999CB01842FC@gmail.com> References: <58C283C2-52C3-48E0-B124-999CB01842FC@gmail.com> Message-ID: Thanks a lot dujinfang !! roberto 2009/3/17 dujinfang : > at default config, in conf/sip_profiles/external.xml > > ? ? > ? ? > ? ? > > where $${external_sip_port} is a variable you can find in conf/ > vars.xml, normally it's 5080, make sure sipbroker route calls to that > port, and then you can make a dialplan in > > conf/dialplan/public.xml > > turn on verbose log in console by > > fs> console loglevel debug > > you can see the process when FS hit the dialplan > > > > On Mar 17, 2009, at 8:26 PM, Roberto Pereyra wrote: > >> Hi all >> >> I'm freswitch newbie ?and have a simple question. >> >> How can enable anonymous inbound calls ? I would like to use >> freeswitch to accept incomming calls from sipbroker DIDs >> >> Any hint ? >> >> Thank in advance for all freeswitch team !! >> >> roberto >> >> -- >> Ing. Roberto Pereyra >> ContenidosOnline >> http://www.contenidosonline.com.ar >> >> The best dedicated servers - LiquidWeb >> http://www.liquidweb.com/?RID=contenid >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 best dedicated servers - LiquidWeb http://www.liquidweb.com/?RID=contenid From benke at inqnet.at Tue Mar 17 08:23:45 2009 From: benke at inqnet.at (Christian Benke) Date: Tue, 17 Mar 2009 16:23:45 +0100 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090311180742.0c6693cb@plex> References: <20090311180742.0c6693cb@plex> Message-ID: <20090317162345.75be1d2e@plex> Hi! Is this not possible with registration at a gateway or is there a other reason why i didn't get any responses on this question? Regards Christian On Wed, 11 Mar 2009 18:07:42 +0100 Christian Benke wrote: > Hi! > > I've recently started to configure a freeswitch for our new office pbx > and so far i like it very much(Coming from asterisk&openser with 2 > years experience at a ITSP. Openser was nice but i didn't like > asterisk for several reasons, so i searched for a more stable and > cleaner alternative. Freeswitch looks _very_ promising and i'd wished > i could use it for more difficult demands than a simple > office-pbx ;-)). > > So far i had little trouble(Though our installation doesn't require > much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. > > The only issue i have not resolved yet is setting the outgoing > DID("head"-number + extension, e.g. +4312345678 + 100). > > The relevant part of the default.xml looks like this atm(where > +4312345678 is our "head"-phone-number without the extensions, > ${caller_id_number} is a 3-digit extension, e.g.: 100): > > data="effective_caller_id_number=+4312345678${caller_id_number}"/> > data="sofia/gateway/sip.myisp.at/${destination_number}"/> > > I'd expect with this dialplan the effective_caller_id would be in the > "From:"-section of the INVITE, but it seems after the bridge it is > overwritten with the gateway-username i've defined in the > gateway-configuration in sip_profiles/external/. > > So instead of: > From: "Desk Phone" > ;tag=U6yQUSta2c2Xg. > i get: > From: "Desk Phone" > ;tag=U6yQUSta2c2Xg. > in the INVITE towards the sip-trunk. > > I may not have grasped yet how proper debugging with freeswitch works, > however, in the console the last action i see, before the bridge to > sofia/external is created, is the setting of the effective-caller-id, > as expected(Do you want to see the whole output?). > > I guess i don't necessarily need to register with the provider, as > they have configured the trunk for my ip-adress and i have theirs in > the ACL(inbound calls work flawless with the head-number+extension), > so maybe the registration is the reason why freeswitch does that > automatically? > > It's probably a little issue, but i don't have the overview yet to > understand how this happens, maybe someone can point me to the right > place? > > Cheers > Christian From mrene_lists at avgs.ca Tue Mar 17 08:26:18 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Mar 2009 11:26:18 -0400 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090311180742.0c6693cb@plex> References: <20090311180742.0c6693cb@plex> Message-ID: gateways have their username in the from section, callerid is sent out as remote-party-id or p-asserted-identity. if you want the from part to have the user you need to set the "caller- id-in-from" param to "true" Math On 11-Mar-09, at 1:07 PM, Christian Benke wrote: > Hi! > > I've recently started to configure a freeswitch for our new office pbx > and so far i like it very much(Coming from asterisk&openser with 2 > years experience at a ITSP. Openser was nice but i didn't like > asterisk > for several reasons, so i searched for a more stable and cleaner > alternative. Freeswitch looks _very_ promising and i'd wished i could > use it for more difficult demands than a simple office-pbx ;-)). > > So far i had little trouble(Though our installation doesn't require > much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. > > The only issue i have not resolved yet is setting the outgoing > DID("head"-number + extension, e.g. +4312345678 + 100). > > The relevant part of the default.xml looks like this atm(where > +4312345678 is our "head"-phone-number without the extensions, > ${caller_id_number} is a 3-digit extension, e.g.: 100): > > data="effective_caller_id_number=+4312345678${caller_id_number}"/> > data="sofia/gateway/sip.myisp.at/${destination_number}"/> > > I'd expect with this dialplan the effective_caller_id would be in the > "From:"-section of the INVITE, but it seems after the bridge it is > overwritten with the gateway-username i've defined in the > gateway-configuration in sip_profiles/external/. > > So instead of: > From: "Desk Phone" > ;tag=U6yQUSta2c2Xg. > i get: > From: "Desk Phone" > ;tag=U6yQUSta2c2Xg. > in the INVITE towards the sip-trunk. > > I may not have grasped yet how proper debugging with freeswitch works, > however, in the console the last action i see, before the bridge to > sofia/external is created, is the setting of the effective-caller- > id, as > expected(Do you want to see the whole output?). > > I guess i don't necessarily need to register with the provider, as > they > have configured the trunk for my ip-adress and i have theirs in > the ACL(inbound calls work flawless with the head-number+extension), > so > maybe the registration is the reason why freeswitch does that > automatically? > > It's probably a little issue, but i don't have the overview yet to > understand how this happens, maybe someone can point me to the right > place? > > Cheers > Christian > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Mar 17 08:27:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 10:27:07 -0500 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090317162345.75be1d2e@plex> References: <20090311180742.0c6693cb@plex> <20090317162345.75be1d2e@plex> Message-ID: <5B965170-FAD4-413C-B38E-FEF0834BE3FB@freeswitch.org> Try export instead of "set" /b On Mar 17, 2009, at 10:23 AM, Christian Benke wrote: >> > data="effective_caller_id_number=+4312345678${caller_id_number}"/> From steveu at coppice.org Tue Mar 17 08:36:52 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 17 Mar 2009 23:36:52 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFBBFD.1050308@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> Message-ID: <49BFC394.6070806@coppice.org> David Knell wrote: > Steve Underwood wrote: >>> When there is Echo being generated from the far end, usually in a >>> bridged call. If you application is just an IVR, with no far end >>> connectivity, then you shouldn't need an echo can. If you are bridging >>> calls, then at some point you may need it, depending on what else is >>> in the loop. >>> >> This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it >> they give very poor reliability detecting DTMF while the prompts are >> playing. If the system uses voice recognition, its reliability will be >> even worse. >> > With respect, this is at best half true. DTMF detection has always > worked just fine > without echo cancellation - the Dialogic, Aculab and Rhetorex cards > which I used > in the late 1990s managed it perfectly well; if the DTMF detection > code in * and FS > can't, then maybe that's something for its author to look at ;-) Try reading the Dialogic and Aculab documentation. Those cards used quite a bit of their DSP capability to remove the spillback of outgoing voice into their DTMF receivers. You'll find the DTMF detector in spandsp (not necessarily the ones in * or FS, which have been altered a bit) is superior to either Dialogic or Aculab's. > ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the > same > hardware above back in the day. I'd be interested to see results of > testing an ASR > engine in with echo; unfortunately, most vendors appear to prohibit > the publication > of test results in their licensing. L&H used to work fine with the J series Dialogic cards. The Dialogic documents go into considerable details about the echo cancellation arrangements to make that happen. Regards, Steve From dujinfang at gmail.com Tue Mar 17 08:44:17 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 17 Mar 2009 23:44:17 +0800 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090317162345.75be1d2e@plex> References: <20090311180742.0c6693cb@plex> <20090317162345.75be1d2e@plex> Message-ID: <620A99C2-26DC-4575-9BF5-47725EAC89EA@gmail.com> Maybe it can help by following this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012083.html On Mar 17, 2009, at 11:23 PM, Christian Benke wrote: > Hi! > > Is this not possible with registration at a gateway or is there a > other > reason why i didn't get any responses on this question? > > Regards > Christian > > On Wed, 11 Mar 2009 18:07:42 +0100 > Christian Benke wrote: > >> Hi! >> >> I've recently started to configure a freeswitch for our new office >> pbx >> and so far i like it very much(Coming from asterisk&openser with 2 >> years experience at a ITSP. Openser was nice but i didn't like >> asterisk for several reasons, so i searched for a more stable and >> cleaner alternative. Freeswitch looks _very_ promising and i'd wished >> i could use it for more difficult demands than a simple >> office-pbx ;-)). >> >> So far i had little trouble(Though our installation doesn't require >> much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. >> >> The only issue i have not resolved yet is setting the outgoing >> DID("head"-number + extension, e.g. +4312345678 + 100). >> >> The relevant part of the default.xml looks like this atm(where >> +4312345678 is our "head"-phone-number without the extensions, >> ${caller_id_number} is a 3-digit extension, e.g.: 100): >> >> > data="effective_caller_id_number=+4312345678${caller_id_number}"/> >> > data="sofia/gateway/sip.myisp.at/${destination_number}"/> >> >> I'd expect with this dialplan the effective_caller_id would be in the >> "From:"-section of the INVITE, but it seems after the bridge it is >> overwritten with the gateway-username i've defined in the >> gateway-configuration in sip_profiles/external/. >> >> So instead of: >> From: "Desk Phone" >> ;tag=U6yQUSta2c2Xg. >> i get: >> From: "Desk Phone" >> ;tag=U6yQUSta2c2Xg. >> in the INVITE towards the sip-trunk. >> >> I may not have grasped yet how proper debugging with freeswitch >> works, >> however, in the console the last action i see, before the bridge to >> sofia/external is created, is the setting of the effective-caller-id, >> as expected(Do you want to see the whole output?). >> >> I guess i don't necessarily need to register with the provider, as >> they have configured the trunk for my ip-adress and i have theirs in >> the ACL(inbound calls work flawless with the head-number+extension), >> so maybe the registration is the reason why freeswitch does that >> automatically? >> >> It's probably a little issue, but i don't have the overview yet to >> understand how this happens, maybe someone can point me to the right >> place? >> >> Cheers >> Christian > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue Mar 17 08:48:28 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Mar 2009 08:48:28 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron wrote: > Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed." > > However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. > Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC From chris at fowler.cc Tue Mar 17 08:49:00 2009 From: chris at fowler.cc (Chris Fowler) Date: Tue, 17 Mar 2009 08:49:00 -0700 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237266076.27059.1305787305@webmail.messagingengine.com> References: <1237266076.27059.1305787305@webmail.messagingengine.com> Message-ID: <1237304940.23519.1305875953@webmail.messagingengine.com> Thanks for the tip Brian. Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log Chris. From anthony.minessale at gmail.com Tue Mar 17 08:53:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Mar 2009 10:53:05 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> Message-ID: <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong > I apologize if this is a double post to -dev. I'm not sure why I don't see > my message appearing, so I'm going to try again in the -user list (first > timer posting here ;). > > I have a situation where it would be useful for a caller not to be hungup, > after finishing the "fifo in" execution (when the agent disconnects the call > or the agent hangs-up). The caller is automatically hungup, in this > situation. It would be preferable if the caller channel went further along > the dial plan. I thought I might get lucky implementing this setting with > hangup_after_bridge to false, but fifo does not utilize this variable. > I tried looking thru the mod_fifo.c source, but my c skills are minimal. I > also tried executing fifo in a lua app and setting setAutoHangup(false), but > that also did not work. Any chance this could be done as a feature > enhancement? Thanks. > > --matt > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/4d60b7fb/attachment.html From brian at freeswitch.org Tue Mar 17 08:53:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 10:53:02 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237304940.23519.1305875953@webmail.messagingengine.com> References: <1237266076.27059.1305787305@webmail.messagingengine.com> <1237304940.23519.1305875953@webmail.messagingengine.com> Message-ID: You're not leaking... I wouldn't call 737 bytes a leak. /b On Mar 17, 2009, at 10:49 AM, Chris Fowler wrote: > Thanks for the tip Brian. > > Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log > > Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/9ee76564/attachment.html From anthony.minessale at gmail.com Tue Mar 17 08:56:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Mar 2009 10:56:50 -0500 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090311180742.0c6693cb@plex> References: <20090311180742.0c6693cb@plex> Message-ID: <191c3a030903170856r15ba2585x6927a00a6983d320@mail.gmail.com> The From: header is not the correct place to place the caller id in SIP yet some providers assume it is. If you add this to your gateway xml config it should fix your problem On Wed, Mar 11, 2009 at 12:07 PM, Christian Benke wrote: > Hi! > > I've recently started to configure a freeswitch for our new office pbx > and so far i like it very much(Coming from asterisk&openser with 2 > years experience at a ITSP. Openser was nice but i didn't like asterisk > for several reasons, so i searched for a more stable and cleaner > alternative. Freeswitch looks _very_ promising and i'd wished i could > use it for more difficult demands than a simple office-pbx ;-)). > > So far i had little trouble(Though our installation doesn't require > much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. > > The only issue i have not resolved yet is setting the outgoing > DID("head"-number + extension, e.g. +4312345678 + 100). > > The relevant part of the default.xml looks like this atm(where > +4312345678 is our "head"-phone-number without the extensions, > ${caller_id_number} is a 3-digit extension, e.g.: 100): > > data="effective_caller_id_number=+4312345678${caller_id_number}"/> > data="sofia/gateway/sip.myisp.at/${destination_number} > "/> > > I'd expect with this dialplan the effective_caller_id would be in the > "From:"-section of the INVITE, but it seems after the bridge it is > overwritten with the gateway-username i've defined in the > gateway-configuration in sip_profiles/external/. > > So instead of: > From: "Desk Phone" > > ;transport=udp>;tag=U6yQUSta2c2Xg. > i get: > From: "Desk Phone" > > ;transport=udp>;tag=U6yQUSta2c2Xg. > in the INVITE towards the sip-trunk. > > I may not have grasped yet how proper debugging with freeswitch works, > however, in the console the last action i see, before the bridge to > sofia/external is created, is the setting of the effective-caller-id, as > expected(Do you want to see the whole output?). > > I guess i don't necessarily need to register with the provider, as they > have configured the trunk for my ip-adress and i have theirs in > the ACL(inbound calls work flawless with the head-number+extension), so > maybe the registration is the reason why freeswitch does that > automatically? > > It's probably a little issue, but i don't have the overview yet to > understand how this happens, maybe someone can point me to the right > place? > > Cheers > Christian > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/53a093f5/attachment-0001.html From anthony.minessale at gmail.com Tue Mar 17 09:03:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Mar 2009 11:03:03 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237304940.23519.1305875953@webmail.messagingengine.com> References: <1237266076.27059.1305787305@webmail.messagingengine.com> <1237304940.23519.1305875953@webmail.messagingengine.com> Message-ID: <191c3a030903170903m41f67916j78f7457af32090c1@mail.gmail.com> The crash on shutdown was an issue in mod_spidermonkey that was accidentally added if you update again it's gone. please run the valgrind command again then make several calls that fall in line with your normal usage pattern so the program can get an accurate trace of the memory usage. On Tue, Mar 17, 2009 at 10:49 AM, Chris Fowler wrote: > Thanks for the tip Brian. > > Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log > > Chris. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/426e6ce9/attachment.html From kzimnicki at gcdf.pl Tue Mar 17 09:24:23 2009 From: kzimnicki at gcdf.pl (Krzysztof Zimnicki) Date: Tue, 17 Mar 2009 17:24:23 +0100 Subject: [Freeswitch-users] Digium TE220P problem. Message-ID: <49BFCEB7.5010703@gcdf.pl> Hi, I have problem with Digium TE220P. Everything works, i can call & talk, but everytime i have CRIT message: 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 When FS start show me ERR message: 2009-03-14 17:44:06 [ERR] Span:0 Q.921() Received UA frame in invalid state This is my config: cat zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span = 2,0,0,ccs,hdb3,crc4 bchan = 32-46,48-62 dchan = 47 loadzone = ru defaultzone=ru cat openzap.conf [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 cat openzap.conf.xml and FS start LOGS: 2009-03-14 17:44:06 [NOTICE] zap_io.c:2612 zap_global_init() Modules configured: 1 2009-03-14 17:44:06 [NOTICE] ozmod_zt.c:922 zt_init() Using Zaptel control device 2009-03-14 17:44:06 [INFO] zap_io.c:2433 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_zt.so [zt] 2009-03-14 17:44:06 [INFO] zap_io.c:2233 load_config() auto-loaded 'zt' 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:38 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 2 as OpenZAP device 1:2 fd:39 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 3 as OpenZAP device 1:3 fd:40 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 4 as OpenZAP device 1:4 fd:41 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 5 as OpenZAP device 1:5 fd:42 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 6 as OpenZAP device 1:6 fd:43 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 7 as OpenZAP device 1:7 fd:44 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 8 as OpenZAP device 1:8 fd:45 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 9 as OpenZAP device 1:9 fd:46 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 10 as OpenZAP device 1:10 fd:47 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 11 as OpenZAP device 1:11 fd:48 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 12 as OpenZAP device 1:12 fd:49 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 13 as OpenZAP device 1:13 fd:50 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 14 as OpenZAP device 1:14 fd:51 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 15 as OpenZAP device 1:15 fd:52 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 16 as OpenZAP device 1:16 fd:53 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 17 as OpenZAP device 1:17 fd:54 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 18 as OpenZAP device 1:18 fd:55 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 19 as OpenZAP device 1:19 fd:56 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 20 as OpenZAP device 1:20 fd:57 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 21 as OpenZAP device 1:21 fd:58 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 22 as OpenZAP device 1:22 fd:59 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 23 as OpenZAP device 1:23 fd:60 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 24 as OpenZAP device 1:24 fd:61 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 25 as OpenZAP device 1:25 fd:62 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 26 as OpenZAP device 1:26 fd:63 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 27 as OpenZAP device 1:27 fd:64 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 28 as OpenZAP device 1:28 fd:65 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 29 as OpenZAP device 1:29 fd:66 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 30 as OpenZAP device 1:30 fd:67 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 31 as OpenZAP device 1:31 fd:68 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 32 as OpenZAP device 2:1 fd:69 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 33 as OpenZAP device 2:2 fd:70 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 34 as OpenZAP device 2:3 fd:71 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 35 as OpenZAP device 2:4 fd:72 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 36 as OpenZAP device 2:5 fd:73 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 37 as OpenZAP device 2:6 fd:74 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 38 as OpenZAP device 2:7 fd:75 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 39 as OpenZAP device 2:8 fd:76 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 40 as OpenZAP device 2:9 fd:77 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 41 as OpenZAP device 2:10 fd:78 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 42 as OpenZAP device 2:11 fd:79 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 43 as OpenZAP device 2:12 fd:80 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 44 as OpenZAP device 2:13 fd:81 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 45 as OpenZAP device 2:14 fd:82 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 46 as OpenZAP device 2:15 fd:83 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 47 as OpenZAP device 2:16 fd:84 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 48 as OpenZAP device 2:17 fd:85 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 49 as OpenZAP device 2:18 fd:86 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 50 as OpenZAP device 2:19 fd:87 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 51 as OpenZAP device 2:20 fd:88 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 52 as OpenZAP device 2:21 fd:89 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 53 as OpenZAP device 2:22 fd:90 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 54 as OpenZAP device 2:23 fd:91 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 55 as OpenZAP device 2:24 fd:92 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 56 as OpenZAP device 2:25 fd:93 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 57 as OpenZAP device 2:26 fd:94 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 58 as OpenZAP device 2:27 fd:95 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 59 as OpenZAP device 2:28 fd:96 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 60 as OpenZAP device 2:29 fd:97 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 61 as OpenZAP device 2:30 fd:98 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 62 as OpenZAP device 2:31 fd:99 2009-03-14 17:44:06 [INFO] zap_io.c:2356 load_config() Configured 62 channel(s) 2009-03-14 17:44:06 [INFO] zap_io.c:2450 zap_load_module() Loading SIG from /usr/local/freeswitch/mod/ozmod_isdn.so 2009-03-14 17:44:06 [INFO] zap_io.c:2566 zap_configure_span() auto-loaded 'isdn' 2009-03-14 17:44:06 [CONSOLE] switch_loadable_module.c:863 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:142 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:240 switch_loadable_module_process() Adding Application 'disable_ec' 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:260 switch_loadable_module_process() Adding API Function 'oz' 2009-03-14 17:44:06 [ERR] Span:0 Q.921() Received UA frame in invalid state When i dial from onference: freeswitch at voipgw> conference test at moh dial openzap/1/3/81048663000000 2009-03-14 17:50:26 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel OpenZAP/1:4/81048663000000 [d6137f1a-10af-11de-8496-c9e9c714d68b] 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 2009-03-14 17:50:30 [NOTICE] mod_openzap.c:1500 on_clear_channel_signal() Ring-Ready OpenZAP/1:4/81048663000000! 2009-03-14 17:50:34 [NOTICE] mod_openzap.c:1491 on_clear_channel_signal() Pre-Answer OpenZAP/1:4/81048663000000! 2009-03-14 17:50:35 [NOTICE] mod_openzap.c:1480 on_clear_channel_signal() Channel [OpenZAP/1:4/81048663000000] has been answered API CALL [conference(test at moh dial openzap/1/4/81048663000000)] output: Call Requested: result: [SUCCESS] 2009-03-14 17:44:06 [CONSOLE] switch_loadable_module.c:863 switch_loadable_module_load_file() Successfully Loaded [mod_commands] freeswitch at voipgw> oz list API CALL [oz(list)] output: +OK span: 1 (span1) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none freeswitch at voipgw> oz dump 1 API CALL [oz(dump 1)] output: +OK span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE span_id: 1 chan_id: 2 physical_span_id: 1 physical_chan_id: 2 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE span_id: 1 chan_id: 3 physical_span_id: 1 physical_chan_id: 3 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE [...] any idea how i can fix this error ? Krzysztof Zimnicki From tntknight at gmail.com Tue Mar 17 09:24:58 2009 From: tntknight at gmail.com (Anthony Knight) Date: Tue, 17 Mar 2009 12:24:58 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFC394.6070806@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> Message-ID: <4cd9d780903170924m79e637d4y4f715891ce16a663@mail.gmail.com> Thanks for the feedback. I have plenty of experience with IVRs and Dialogic cards (starting with D121/LSI120s and SS96s under DOS in the 90's all the way up to Intel's DM/Vs) and didn't ever have a problem with DTMF collection with ISDN PRI lines except occasionally with wireless and cell phones (Bad line quality). These new cards are so much cheaper than the Dialogic cards were, I should just buy the version with the cancellers. Tony On Tue, Mar 17, 2009 at 11:36 AM, Steve Underwood wrote: > David Knell wrote: > > Steve Underwood wrote: > >>> When there is Echo being generated from the far end, usually in a > >>> bridged call. If you application is just an IVR, with no far end > >>> connectivity, then you shouldn't need an echo can. If you are bridging > >>> calls, then at some point you may need it, depending on what else is > >>> in the loop. > >>> > >> This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it > >> they give very poor reliability detecting DTMF while the prompts are > >> playing. If the system uses voice recognition, its reliability will be > >> even worse. > >> > > With respect, this is at best half true. DTMF detection has always > > worked just fine > > without echo cancellation - the Dialogic, Aculab and Rhetorex cards > > which I used > > in the late 1990s managed it perfectly well; if the DTMF detection > > code in * and FS > > can't, then maybe that's something for its author to look at ;-) > Try reading the Dialogic and Aculab documentation. Those cards used > quite a bit of their DSP capability to remove the spillback of outgoing > voice into their DTMF receivers. You'll find the DTMF detector in > spandsp (not necessarily the ones in * or FS, which have been altered a > bit) is superior to either Dialogic or Aculab's. > > ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the > > same > > hardware above back in the day. I'd be interested to see results of > > testing an ASR > > engine in with echo; unfortunately, most vendors appear to prohibit > > the publication > > of test results in their licensing. > L&H used to work fine with the J series Dialogic cards. The Dialogic > documents go into considerable details about the echo cancellation > arrangements to make that happen. > > Regards, > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/2458886b/attachment.html From benke at inqnet.at Tue Mar 17 09:38:35 2009 From: benke at inqnet.at (Christian Benke) Date: Tue, 17 Mar 2009 17:38:35 +0100 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <191c3a030903170856r15ba2585x6927a00a6983d320@mail.gmail.com> References: <20090311180742.0c6693cb@plex> <191c3a030903170856r15ba2585x6927a00a6983d320@mail.gmail.com> Message-ID: <20090317173835.68ee5df3@plex> wow, now that was fast :-) Cheers for all replies, setting the caller-id-in-from-parameter was sufficient! regards Christian From steveu at coppice.org Tue Mar 17 09:38:55 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 00:38:55 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <4cd9d780903170924m79e637d4y4f715891ce16a663@mail.gmail.com> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <4cd9d780903170924m79e637d4y4f715891ce16a663@mail.gmail.com> Message-ID: <49BFD21F.3060303@coppice.org> Anthony Knight wrote: > Thanks for the feedback. > > I have plenty of experience with IVRs and Dialogic cards (starting > with D121/LSI120s and SS96s under DOS in the 90's all the way up to > Intel's DM/Vs) and didn't ever have a problem with DTMF collection > with ISDN PRI lines except occasionally with wireless and cell phones > (Bad line quality). You have my deepest sympathy. :-) Steve From msc at freeswitch.org Tue Mar 17 09:42:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Mar 2009 09:42:05 -0700 Subject: [Freeswitch-users] [OpenZap] problem with TE220P In-Reply-To: <49BF6A99.6070308@gcdf.pl> References: <49BF6A99.6070308@gcdf.pl> Message-ID: <87f2f3b90903170942m3ae6bf05x1a827ec49efcf46a@mail.gmail.com> > any idea how i can fix this error ? > I believe this is a harmless warning. However, you might try to use ozmod_libpri, which uses the libpri PRI stack instead of the built-in OpenZAP PRI stack. More info here: http://wiki.freeswitch.org/wiki/OpenZAP#OpenZAP_Installation -MC From dave at 3c.co.uk Tue Mar 17 10:46:27 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Mar 2009 11:46:27 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFC394.6070806@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> Message-ID: <49BFE1F3.2030207@3c.co.uk> Steve Underwood wrote: > David Knell wrote: > >> Steve Underwood wrote: >> >>>> When there is Echo being generated from the far end, usually in a >>>> bridged call. If you application is just an IVR, with no far end >>>> connectivity, then you shouldn't need an echo can. If you are bridging >>>> calls, then at some point you may need it, depending on what else is >>>> in the loop. >>>> >>>> >>> This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it >>> they give very poor reliability detecting DTMF while the prompts are >>> playing. If the system uses voice recognition, its reliability will be >>> even worse. >>> >>> >> With respect, this is at best half true. DTMF detection has always >> worked just fine >> without echo cancellation - the Dialogic, Aculab and Rhetorex cards >> which I used >> in the late 1990s managed it perfectly well; if the DTMF detection >> code in * and FS >> can't, then maybe that's something for its author to look at ;-) >> > Try reading the Dialogic and Aculab documentation. Those cards used > quite a bit of their DSP capability to remove the spillback of outgoing > voice into their DTMF receivers. You'll find the DTMF detector in > spandsp (not necessarily the ones in * or FS, which have been altered a > bit) is superior to either Dialogic or Aculab's. > The first bit of that's a tad patronising, isn't it, and, in the case of the decade-old Aculab cards which which I'm most familiar, is also untrue. As for the second, do you have any test results to back that up? I'm more curious than setting out for an argument.. >> ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the >> same >> hardware above back in the day. I'd be interested to see results of >> testing an ASR >> engine in with echo; unfortunately, most vendors appear to prohibit >> the publication >> of test results in their licensing. >> > L&H used to work fine with the J series Dialogic cards. The Dialogic > documents go into considerable details about the echo cancellation > arrangements to make that happen. > > You've missed the point I was trying to make. It used to work fine with no echo cancellation at all. --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/5ccc6fae/attachment.html From ludovic.fouquet at bewan.com Tue Mar 17 11:16:33 2009 From: ludovic.fouquet at bewan.com (ludovic) Date: Tue, 17 Mar 2009 19:16:33 +0100 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: References: <49BE928D.3090509@bewan.com> <49BF87E5.5090809@bewan.com> Message-ID: <49BFE901.3070709@bewan.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/faa72201/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: bewan100.jpg Type: image/jpeg Size: 3963 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/faa72201/attachment-0001.jpg From brian at freeswitch.org Tue Mar 17 11:26:31 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 13:26:31 -0500 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: <49BFE901.3070709@bewan.com> References: <49BE928D.3090509@bewan.com> <49BF87E5.5090809@bewan.com> <49BFE901.3070709@bewan.com> Message-ID: <246296A5-851A-4859-BCA9-05E2415A20EA@freeswitch.org> I checked, I don't see sw1.freephonie.net in the logs trying to resolve it... and the SRV records are all correct as are the naptr records which is shocking ;) /b On Mar 17, 2009, at 1:16 PM, ludovic wrote: > I understood the example. > What I mean is that my DNS issue comes from sofia-sip and my sip > provider (freephonie.net) name resolution which fails when calling > whereas it is well resolved during the registration process. > Here is a trace : > 2009-03-17 18:47:28 [NOTICE] switch_channel.c:538 > switch_channel_set_name() New Channel sofia/external/0123456789 > [ae9c2108-131b-11de-8a2f-1fafbaa54120] nua: > nh_create_handle: entering > ; > nua: nua_handle_bind: entering > nua: nua_invite: entering > nua: nua_stack_set_params: entering > soa_clone(static::0x100abd98, 0x100a8738, 0x100c6748) called > soa_set_params(static::0x100b4728, ...) called > soa_set_params(static::0x100b4728, ...) called > soa_set_user_sdp(static::0x100b4728, (nil), 0x10106364, -1) called > soa_set_capability_sdp(static::0x100b4728, (nil), 0x10106364, -1) > called > su_localinfo: if lo with index 1 > su_localinfo: if lan1 with index 18 > su_localinfo: if ppp1 with index 23 > nta_leg_tcreate(0x100c7f30) > nua(0x100c6748): adding session usage > soa_init_offer_answer(static::0x100b4728) called > soa_generate_offer(static::0x100b4728, 0) called > soa_static_offer_answer_action(0x100b4728, soa_generate_offer): called > soa_static(0x100b4728, soa_generate_offer): generating local > description > su_localinfo: if lo with index 1 > su_localinfo: if lan1 with index 18 > su_localinfo: if ppp1 with index 23 > soa_static(0x100b4728, soa_generate_offer): upgrade with local > description > soa_sdp_mode_set(0x7dbfd850, (nil), ""): called > soa_static(0x100b4728, soa_generate_offer): storing local description > soa_get_local_sdp(static::0x100b4728, [(nil)], [0x7dbff980], > [0x7dbff984]) called > nta: selecting scheme sip > sres_cache_get(0x100abfc0, NAPTR, "freephonie.net.") called > rr found in cache: freephonie.net. 35 > sres_cache_get(0x100abfc0, NAPTR, "freephonie.net.") returned 1 > entries > nta: for "freephonie.net" query "freephonie.net" NAPTR (cached) > nta: freephonie.net. IN NAPTR 100 100 "s" "SIP+D2U" "" > _sip_udp.freephonie.net. > sres_cache_get(0x100abfc0, SRV, "_sip_udp.freephonie.net.") called > rr found in cache: _sip_udp.freephonie.net. 33 > sres_cache_get(0x100abfc0, SRV, "_sip_udp.freephonie.net.") returned > 1 entries > nta: for "freephonie.net" query "_sip_udp.freephonie.net." SRV > (cached) > nta: timer set to 32000 ms > nua(0x100c6748): call state changed: init -> calling, sent offer > soa_get_local_sdp(static::0x100b4728, [0x7dbff988], [0x7dbff98c], > [(nil)]) called > nua: nua_application_event: entering > nua(0x100c6748): call state changed: calling -> init > nua(0x100c6748): removing session usage > soa_destroy(static::0x100b4728) called From steveu at coppice.org Tue Mar 17 11:28:27 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 02:28:27 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFE1F3.2030207@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> Message-ID: <49BFEBCB.9020708@coppice.org> David Knell wrote: > Steve Underwood wrote: >> David Knell wrote: >> >>> Steve Underwood wrote: >>> >>>>> When there is Echo being generated from the far end, usually in a >>>>> bridged call. If you application is just an IVR, with no far end >>>>> connectivity, then you shouldn't need an echo can. If you are bridging >>>>> calls, then at some point you may need it, depending on what else is >>>>> in the loop. >>>>> >>>>> >>>> This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it >>>> they give very poor reliability detecting DTMF while the prompts are >>>> playing. If the system uses voice recognition, its reliability will be >>>> even worse. >>>> >>>> >>> With respect, this is at best half true. DTMF detection has always >>> worked just fine >>> without echo cancellation - the Dialogic, Aculab and Rhetorex cards >>> which I used >>> in the late 1990s managed it perfectly well; if the DTMF detection >>> code in * and FS >>> can't, then maybe that's something for its author to look at ;-) >>> >> Try reading the Dialogic and Aculab documentation. Those cards used >> quite a bit of their DSP capability to remove the spillback of outgoing >> voice into their DTMF receivers. You'll find the DTMF detector in >> spandsp (not necessarily the ones in * or FS, which have been altered a >> bit) is superior to either Dialogic or Aculab's. >> > The first bit of that's a tad patronising, isn't it, You are the one who started out being offensive. > and, in the case of the decade-old Aculab > cards which which I'm most familiar, is also untrue. I can't find too much about the old cards on the web now, but I found http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html which is pretty much a copy and paste from the old Dialogic web pages, and you'll see it says "Cut through : Local echo cancellation permits 100% detection with a >4.5 dB return loss line". The Aculabs did the same thing for sure. They just couldn't work without cancellation. There were some very early Dialogic cards, using DTMF receiver chips and OKI ADPCM chips, and had no general purpose DSPs. They performed really badly because of the lack of cancellation, and were quickly replaced with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms into a Motorola 56k DSP chips. > > As for the second, do you have any test results to back that up? I'm > more curious than > setting out for an argument.. >>> ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the >>> same >>> hardware above back in the day. I'd be interested to see results of >>> testing an ASR >>> engine in with echo; unfortunately, most vendors appear to prohibit >>> the publication >>> of test results in their licensing. >>> >> L&H used to work fine with the J series Dialogic cards. The Dialogic >> documents go into considerable details about the echo cancellation >> arrangements to make that happen. >> >> > You've missed the point I was trying to make. It used to work fine > with no echo cancellation > at all. I think you've missed the point. These things don't work by pixey dust. They work by engineering. If you have any old J or JCT cards around record the signal from the far end. You'll find only the tiniest trace of the outgoing signal mixed in with it. How do you think that happens? Steve From jp.manchu at gmail.com Tue Mar 17 13:48:18 2009 From: jp.manchu at gmail.com (JayaPrakash) Date: Wed, 18 Mar 2009 02:18:18 +0530 Subject: [Freeswitch-users] Nibblebill - DB Error while updating cash! Message-ID: Hi All, I have installed nibblebill and* it is able to bill the calls.* However, it is giving following error in FreeSwitch server. 2009-03-17 23:17:19 [DEBUG] mod_nibblebill.c:283 bill_event() Doing update query [UPDATE accounts SET cash=cash-0.045767 WHERE id="1"] 2009-03-17 23:17:19 [CRIT] mod_nibblebill.c:286 bill_event() DB Error while updating cash! Thanks Jayaprakash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/f95a7def/attachment.html From ctalle at voiceway.ca Tue Mar 17 14:13:16 2009 From: ctalle at voiceway.ca (Cristian Talle) Date: Tue, 17 Mar 2009 17:13:16 -0400 Subject: [Freeswitch-users] DTMF detection during bridge Message-ID: <49C0126C.2080107@voiceway.ca> Hi, Is there any easy way to get in FS the same behavior as when using the "d" flag with asterisk's Dial command? I need FS to jump to a different extension if the caller presses a digit while waiting for the called party to answer. *"...d*: intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered..." Thanks, Cristian From brian at freeswitch.org Tue Mar 17 14:24:53 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 16:24:53 -0500 Subject: [Freeswitch-users] DTMF detection during bridge In-Reply-To: <49C0126C.2080107@voiceway.ca> References: <49C0126C.2080107@voiceway.ca> Message-ID: Check out the bind_meta_app that exists in the default examples... I think thats what you want. /b On Mar 17, 2009, at 4:13 PM, Cristian Talle wrote: > Hi, > > Is there any easy way to get in FS the same behavior as when using the > "d" flag with asterisk's Dial command? > I need FS to jump to a different extension if the caller presses a > digit > while waiting for the called party to answer. > > *"...d*: intercepts any dtmf while waiting for the call to be answered > and returns that value on the spot. This allows you to dial a 1-digit > exit extension while waiting for the call to be answered..." > > Thanks, > Cristian From mthomas at themarketbuilder.com Tue Mar 17 15:35:48 2009 From: mthomas at themarketbuilder.com (Mark Thomas) Date: Tue, 17 Mar 2009 15:35:48 -0700 Subject: [Freeswitch-users] Newbie question: Why can't I dial? Message-ID: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> Hello, everyone. I am new to Freeswitch, and telephony in general. I am trying to set up a Freeswitch system at work for a project, and I have hit a wall. I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I believe I have openzap correctly installed in wanpipe mode. I am trying to bridge an incoming SIP call from an IP phone to an openzap channel without success. The Freeswitch log shows that dialing takes place, but the call never completes. The call log is here: http://pastebin.freeswitch.org/7805 The dialplan xml, openzap.conf, and openzap.conf.xml are here: http://pastebin.freeswitch.org/7806 Any help greatly appreciated. --Mark From brian at freeswitch.org Tue Mar 17 15:54:47 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 17:54:47 -0500 Subject: [Freeswitch-users] Newbie question: Why can't I dial? In-Reply-To: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> References: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> Message-ID: Mark, You should join both #openzap and #freeswitch on irc.freenode.net there are way too many things to go over and the list would just be too slow. /b On Mar 17, 2009, at 5:35 PM, Mark Thomas wrote: > Hello, everyone. > > I am new to Freeswitch, and telephony in general. I am trying to > set up a Freeswitch system at work for a project, and I have hit a > wall. > > I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I > believe I have openzap correctly installed in wanpipe mode. I am > trying to bridge an incoming SIP call from an IP phone to an openzap > channel without success. The Freeswitch log shows that dialing > takes place, but the call never completes. > > The call log is here: http://pastebin.freeswitch.org/7805 > The dialplan xml, openzap.conf, and openzap.conf.xml are here: http://pastebin.freeswitch.org/7806 > > Any help greatly appreciated. > > --Mark > > ______________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/793c969f/attachment.html From msc at freeswitch.org Tue Mar 17 16:17:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Mar 2009 16:17:05 -0700 Subject: [Freeswitch-users] Newbie question: Why can't I dial? In-Reply-To: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> References: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> Message-ID: <87f2f3b90903171617l266db132vf9143c4562cad407@mail.gmail.com> On Tue, Mar 17, 2009 at 3:35 PM, Mark Thomas wrote: > Hello, everyone. > > I am new to Freeswitch, and telephony in general. ?I am trying to set up a Freeswitch system at work for a project, and I have hit a wall. > > I have a dedicated LD T1 from Qwest and a Sangoma A104 card. ?I believe I have openzap correctly installed in wanpipe mode. I am trying to bridge an incoming SIP call from an IP phone to an openzap channel without success. ?The Freeswitch log shows that dialing takes place, but the call never completes. > > The call log is here: http://pastebin.freeswitch.org/7805 > The dialplan xml, openzap.conf, and openzap.conf.xml are here: http://pastebin.freeswitch.org/7806 > > Any help greatly appreciated. > Actually I found two things you need to change in the dialplan. What's happening is that you are telling openzap to dial out span 1, lowest channel number, but you don't actually give it a phone number to dial. Here's the current dialplan: first, your expression is a bit dangerous. second, it doesn't actually "capture" the dialed number. I recommend that you do something like this: Note the leading nine, the \d+ and the parentheses. Essentially this regex says: Match any string of digits that begins with a 9 and has at least one additional digit. The parens will put the value of (\d+) into the variable $1. Your bridge then would be this: Now, reload your dialplan (press F6 or type "reloadxml" at the CLI) and dial out with a leading 9: 95551212 will send 5551212 to the telco. Try it and report back! -MC (IRC: mercutioviz) > --Mark > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dave at 3c.co.uk Tue Mar 17 16:21:11 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Mar 2009 17:21:11 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFEBCB.9020708@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> Message-ID: <49C03067.7070406@3c.co.uk> Steve Underwood wrote: > [whopping big snip] > >> The first bit of that's a tad patronising, isn't it, >> > You are the one who started out being offensive. > I'm sorry if you find disagreement offensive; you might not wish to read beyond this point if so. >> and, in the case of the decade-old Aculab >> cards which which I'm most familiar, is also untrue. >> > I can't find too much about the old cards on the web now, but I found > http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html > which is pretty much a copy and paste from the old Dialogic web pages, > and you'll see it says "Cut through : Local echo cancellation permits > 100% detection with a >4.5 dB return loss line". The Aculabs did the > same thing for sure. They just couldn't work without cancellation. There > were some very early Dialogic cards, using DTMF receiver chips and OKI > ADPCM chips, and had no general purpose DSPs. They performed really > badly because of the lack of cancellation, and were quickly replaced > with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms > into a Motorola 56k DSP chips. > The same document, under the bit which you've quoted, says: "(E-1) Digital trunks use separate transmit and receive paths to network. Performance dependent on far end handset's match to local analog loop." - i.e. the card does no echo cancellation. Aculab didn't even offer echo cancellation on Prosody for years and, when they did, it consumed prodigious amounts of DSP. Nonetheless, the DTMF detection worked perfectly well, even across 120 channels per 40MHz SHARC - there's just no way that those DSPs had enough horsepower to do echo cancellation across that many channels. An Asterisk box with an el-cheapo quad E1 card in that I use for TDM-SIP gatewaying detects DTMF perfectly well with no echo cancellation. You just don't need echo cancellation to achieve perfectly acceptable DTMF detection. ASR - yes, maybe, but surely only in the case where the application requires barge-in; even then, I'd be interested to see some test results, particuarly where the outbound prompt is killed the moment the ASR reports start of speech. I'm afraid that your original bald claim - that "IVRs badly need echo cancellation" is simply wrong, misleading and irresponsible: those believing it will end up spending large sums of money on technology which they probably do not need. --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/a3e1397f/attachment-0001.html From msc at freeswitch.org Tue Mar 17 16:40:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Mar 2009 16:40:38 -0700 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C03067.7070406@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> Message-ID: <87f2f3b90903171640i25da1894xa062d4f91450e97f@mail.gmail.com> > I'm afraid that your original bald claim - that "IVRs badly need echo > cancellation" is simply > wrong, misleading and irresponsible: those believing it will end up spending > large sums > of money on technology which they probably do not need. Anybody with years, perhaps decades, of DSP programming experience plus testing in the real world - and all over the world - has my vote of confidence. Furthermore, when this person writes spandsp, makes it open source, and freely answers questions about it on public fora, I am inclined not only to believe him but to trust his judgment. Bottom line: thousands of people have chosen to heed Steve's advice. He is well-respected in many technical communities. His reputation is as solid as it gets. "Do what Steve says" is about the safest bet you will ever make in this business. -MC From steveu at coppice.org Tue Mar 17 17:25:01 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 08:25:01 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C03067.7070406@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> Message-ID: <49C03F5D.9050904@coppice.org> David Knell wrote: > Steve Underwood wrote: >> [whopping big snip] >> >>> The first bit of that's a tad patronising, isn't it, >>> >> You are the one who started out being offensive. >> > I'm sorry if you find disagreement offensive; you might not wish to > read beyond this > point if so. >>> and, in the case of the decade-old Aculab >>> cards which which I'm most familiar, is also untrue. >>> >> I can't find too much about the old cards on the web now, but I found >> http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html >> which is pretty much a copy and paste from the old Dialogic web pages, >> and you'll see it says "Cut through : Local echo cancellation permits >> 100% detection with a >4.5 dB return loss line". The Aculabs did the >> same thing for sure. They just couldn't work without cancellation. There >> were some very early Dialogic cards, using DTMF receiver chips and OKI >> ADPCM chips, and had no general purpose DSPs. They performed really >> badly because of the lack of cancellation, and were quickly replaced >> with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms >> into a Motorola 56k DSP chips. >> > The same document, under the bit which you've quoted, says: > "(E-1) Digital trunks use separate transmit and receive paths to network. > Performance dependent on far end handset's match to local analog loop." > - i.e. the card does no echo cancellation. Your messages are starting to looked deranged. Why would they only apply echo cancellation to T1s? Its a bizarre idea, and you must realise its wrong. Are you so desperate to support a wrong answer you'll clutch at straws? :-\ > > Aculab didn't even offer echo cancellation on Prosody for years and, > when they did, it > consumed prodigious amounts of DSP. Nonetheless, the DTMF detection > worked > perfectly well, even across 120 channels per 40MHz SHARC - there's > just no way > that those DSPs had enough horsepower to do echo cancellation across > that many > channels. This page http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html seems to support what you say. It also implies DTMF detection sucks unless you echo cancel. The statement "If the outgoing signal is a tone of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" is telling you to keep your outgoing signal in the same frequency range as dial-tone where the dial-tone filter on the DTMF receiver will obviate the need for an echo canceller. They are freely admitting exactly what I have said. If you want a normal IVR with cut-through to work you better turn that echo canceller on. My only experience with Aculab was fitting a box designed by other people into a system. That one definitely echo cancelled, as it worked as well as the Dialogic based boxes we developed ourselves. > > An Asterisk box with an el-cheapo quad E1 card in that I use for > TDM-SIP gatewaying > detects DTMF perfectly well with no echo cancellation. You must have very low standards for "works well". > > You just don't need echo cancellation to achieve perfectly acceptable > DTMF detection. Well, not if you expect people to wait for silence before entering DTMF, but who would tolerate that these days? Cut through has been de rigeur since the late 80s. > > ASR - yes, maybe, but surely only in the case where the application > requires barge-in; > even then, I'd be interested to see some test results, particuarly > where the outbound prompt > is killed the moment the ASR reports start of speech. Doesn't any sane system expect barge in to be nearly as reliable as waiting for silence? Who would tolerate something that doesn't? It has been a standard expectation of any decent IVR since they began. > > I'm afraid that your original bald claim - that "IVRs badly need echo > cancellation" is simply > wrong, misleading and irresponsible: those believing it will end up > spending large sums > of money on technology which they probably do not need. You must have very low standards for what works well. If you suggest people leave out echo cancellation you are just asking for customer service issues down the line. That whole Aculab page is a clear response to just such issues they had, which forced them to add the necessary improvements. Regards, Steve From d at d-man.org Tue Mar 17 18:31:04 2009 From: d at d-man.org (Darren Schreiber) Date: Tue, 17 Mar 2009 18:31:04 -0700 Subject: [Freeswitch-users] Nibblebill - DB Error while updating cash! In-Reply-To: References: Message-ID: Hi there, The updates to the DB are working, but the error is still being thrown. I will try and fix this tonight. Diego also reported the same issue last week, I just haven't gotten around to it. My apologies. The bug is filed, I'll close it out when it's fixed. - Darren _____ From: JayaPrakash [mailto:jp.manchu at gmail.com] Sent: Tuesday, March 17, 2009 1:48 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Nibblebill - DB Error while updating cash! Hi All, I have installed nibblebill and it is able to bill the calls. However, it is giving following error in FreeSwitch server. 2009-03-17 23:17:19 [DEBUG] mod_nibblebill.c:283 bill_event() Doing update query [UPDATE accounts SET cash=cash-0.045767 WHERE id="1"] 2009-03-17 23:17:19 [CRIT] mod_nibblebill.c:286 bill_event() DB Error while updating cash! Thanks Jayaprakash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/7190920b/attachment.html From jason at jasonjgw.net Tue Mar 17 18:46:59 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 18 Mar 2009 12:46:59 +1100 Subject: [Freeswitch-users] TLS support in Debian build Message-ID: <20090318014659.GA15840@jdc.jasonjgw.net> I've just tried enabling TLS support, and the SIP profiles with TLS enabled in them won't load. According to the wiki, this is typically the result of missing headers during the build process, with TLS having not been included. However, on my Debian system, I have header files under /usr/include/openssl, which come from the libssl-dev package. Thus, SSL/TLS should have been compiled into FreeSWITCH unless there's something amiss with the Debian build process. Suggestions for tracking this down are welcome. There's no urgency: this is just for experimental purposes, after all. From brian at freeswitch.org Tue Mar 17 18:53:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 20:53:58 -0500 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: <20090318014659.GA15840@jdc.jasonjgw.net> References: <20090318014659.GA15840@jdc.jasonjgw.net> Message-ID: <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> if you installed the ssl devel stuff AFTER you configured you'll need to reconfigure. /b On Mar 17, 2009, at 8:46 PM, Jason White wrote: > I've just tried enabling TLS support, and the SIP profiles with TLS > enabled in > them won't load. > > According to the wiki, this is typically the result of missing > headers during > the build process, with TLS having not been included. > > However, on my Debian system, I have header files under /usr/include/ > openssl, > which come from the libssl-dev package. Thus, SSL/TLS should have been > compiled into FreeSWITCH unless there's something amiss with the > Debian build > process. > > Suggestions for tracking this down are welcome. There's no urgency: > this is > just for experimental purposes, after all. From dave at 3c.co.uk Tue Mar 17 19:02:41 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Mar 2009 20:02:41 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C03F5D.9050904@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> Message-ID: <49C05641.7070309@3c.co.uk> Steve Underwood wrote: > David Knell wrote: > >> Steve Underwood wrote: >> >>> [whopping big snip] >>> >>> >>>> The first bit of that's a tad patronising, isn't it, >>>> >>>> >>> You are the one who started out being offensive. >>> >>> >> I'm sorry if you find disagreement offensive; you might not wish to >> read beyond this >> point if so. >> >>>> and, in the case of the decade-old Aculab >>>> cards which which I'm most familiar, is also untrue. >>>> >>>> >>> I can't find too much about the old cards on the web now, but I found >>> http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html >>> which is pretty much a copy and paste from the old Dialogic web pages, >>> and you'll see it says "Cut through : Local echo cancellation permits >>> 100% detection with a >4.5 dB return loss line". The Aculabs did the >>> same thing for sure. They just couldn't work without cancellation. There >>> were some very early Dialogic cards, using DTMF receiver chips and OKI >>> ADPCM chips, and had no general purpose DSPs. They performed really >>> badly because of the lack of cancellation, and were quickly replaced >>> with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms >>> into a Motorola 56k DSP chips. >>> >>> >> The same document, under the bit which you've quoted, says: >> "(E-1) Digital trunks use separate transmit and receive paths to network. >> Performance dependent on far end handset's match to local analog loop." >> - i.e. the card does no echo cancellation. >> > Your messages are starting to looked deranged. Why would they only apply > echo cancellation to T1s? Its a bizarre idea, and you must realise its > wrong. Are you so desperate to support a wrong answer you'll clutch at > straws? :-\ > More insults. Answer me this: if there were echo cancellation in use, why would DTMF detection performance depend on the far-end handset's match to the loop? And the follow-up question (which you've already pretty much asked) - if the card doesn't echo cancel for E1s, why would it for T1s? As an aside, I'm not convinced that the document's not talking about return loss on the T1 line itself, the implication being that the T1 is being carried on a single pair, which makes the first sentence about E1s make a bit more sense. But that's just a guess. >> Aculab didn't even offer echo cancellation on Prosody for years and, >> when they did, it >> consumed prodigious amounts of DSP. Nonetheless, the DTMF detection >> worked >> perfectly well, even across 120 channels per 40MHz SHARC - there's >> just no way >> that those DSPs had enough horsepower to do echo cancellation across >> that manychannels. >> > This page > http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html > seems to support what you say. It also implies DTMF detection sucks > unless you echo cancel. The statement "If the outgoing signal is a tone > of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" > is telling you to keep your outgoing signal in the same frequency range > as dial-tone where the dial-tone filter on the DTMF receiver will > obviate the need for an echo canceller. They are freely admitting > exactly what I have said. If you want a normal IVR with cut-through to > work you better turn that echo canceller on. > > My only experience with Aculab was fitting a box designed by other > people into a system. That one definitely echo cancelled, as it worked > as well as the Dialogic based boxes we developed ourselves. > That only holds true if your premise - that you need echo cancellation for good DTMF detection - is correct, which I don't believe it is. >> An Asterisk box with an el-cheapo quad E1 card in that I use for >> TDM-SIP gatewaying >> detects DTMF perfectly well with no echo cancellation. >> > You must have very low standards for "works well". > Nothing like a good old ad hominem attack. Beats reasoned argument any day. >> You just don't need echo cancellation to achieve perfectly acceptable >> DTMF detection. >> > Well, not if you expect people to wait for silence before entering DTMF, > but who would tolerate that these days? Cut through has been de rigeur > since the late 80s. > Oh, for pity's sake, you get perfectly good cut through without echo cancellation. Humour me and draw a quick mental picture of the spectrum of a random bit of speech at -20dBm; now add tones at -10dBm and -7dBm. They stick out like a pair of sore thumbs. I'm sure it's quite possible to come up with a pathological case - e.g. cut-through against a 1kHz milliwatt tone, but that sort of thing just doesn't happen in real- life IVR applications. >> ASR - yes, maybe, but surely only in the case where the application >> requires barge-in; >> even then, I'd be interested to see some test results, particuarly >> where the outbound prompt >> is killed the moment the ASR reports start of speech. >> > Doesn't any sane system expect barge in to be nearly as reliable as > waiting for silence? Who would tolerate something that doesn't? It has > been a standard expectation of any decent IVR since they began. > Sorry - ASR with barge-in has been a standard expectation since the first IVRs? >> I'm afraid that your original bald claim - that "IVRs badly need echo >> cancellation" is simply >> wrong, misleading and irresponsible: those believing it will end up >> spending large sums >> of money on technology which they probably do not need. >> > You must have very low standards for what works well. If you suggest > people leave out echo cancellation you are just asking for customer > service issues down the line. That whole Aculab page is a clear response > to just such issues they had, which forced them to add the necessary > improvements. > Repeating you ad-hominem really doesn't make it any stronger, I'm afraid. And the Aculab page you refer to offers four solutions for problems caused by far- end echo, of which cancellation is just one; not playing a stationary tone above 600Hz is another. Do you have any real-world samples of DTMF+echo which give your DTMF detection code trouble? --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/0ed589c6/attachment-0001.html From jason at jasonjgw.net Tue Mar 17 19:31:54 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 18 Mar 2009 13:31:54 +1100 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> References: <20090318014659.GA15840@jdc.jasonjgw.net> <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> Message-ID: <20090318023154.GA16523@jdc.jasonjgw.net> Brian West wrote: > if you installed the ssl devel stuff AFTER you configured you'll need > to reconfigure. I'm reasonably sure it was installed already, unless it was pulled in recently by a package upgrade. The configure script needs to look in /usr/include/openssl for the headers. I'll have a look at config.log and try to work out what it looked for and why it didn't find it. From egghunt at gmail.com Tue Mar 17 19:40:59 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Tue, 17 Mar 2009 23:40:59 -0300 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C05641.7070309@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> Message-ID: Sharing my humble experience: in Brazil we usually need echo cancellation to have reliable DTMF detection _and_ voice quality over E1 lines (be it on MFC/R2 - r2d - or ISDN PRI lines), either for sip/tdm gateway devices or IVR applications. Usually there's no need for echo cancellation on links from some Telcos, in some specific places. But we need it in the majority of cases, even when my box is just a gateway between legacy pbxes. This represents just a subset of the available E1s in the world and it's just a practical experience, but it's a fact for me. If I don't have a card with echo cancellation, I don't offer reliability to my customer; I've done that in the past and didn't work out. I'm not theoretically discussing anything, just sharing what I've been through in the last 4 or 5 years. 2009/3/17 David Knell > Steve Underwood wrote: > > David Knell wrote: > > > Steve Underwood wrote: > > > [whopping big snip] > > > > The first bit of that's a tad patronising, isn't it, > > > > You are the one who started out being offensive. > > > > I'm sorry if you find disagreement offensive; you might not wish to > read beyond this > point if so. > > > and, in the case of the decade-old Aculab > cards which which I'm most familiar, is also untrue. > > > > I can't find too much about the old cards on the web now, but I found http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html > which is pretty much a copy and paste from the old Dialogic web pages, > and you'll see it says "Cut through : Local echo cancellation permits > 100% detection with a >4.5 dB return loss line". The Aculabs did the > same thing for sure. They just couldn't work without cancellation. There > were some very early Dialogic cards, using DTMF receiver chips and OKI > ADPCM chips, and had no general purpose DSPs. They performed really > badly because of the lack of cancellation, and were quickly replaced > with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms > into a Motorola 56k DSP chips. > > > > The same document, under the bit which you've quoted, says: > "(E-1) Digital trunks use separate transmit and receive paths to network. > Performance dependent on far end handset's match to local analog loop." > - i.e. the card does no echo cancellation. > > > Your messages are starting to looked deranged. Why would they only apply > echo cancellation to T1s? Its a bizarre idea, and you must realise its > wrong. Are you so desperate to support a wrong answer you'll clutch at > straws? :-\ > > > More insults. Answer me this: if there were echo cancellation in use, why > would > DTMF detection performance depend on the far-end handset's match to the > loop? > > And the follow-up question (which you've already pretty much asked) - if > the > card doesn't echo cancel for E1s, why would it for T1s? > > As an aside, I'm not convinced that the document's not talking about return > loss > on the T1 line itself, the implication being that the T1 is being carried > on a single > pair, which makes the first sentence about E1s make a bit more sense. But > that's > just a guess. > > Aculab didn't even offer echo cancellation on Prosody for years and, > when they did, it > consumed prodigious amounts of DSP. Nonetheless, the DTMF detection > worked > perfectly well, even across 120 channels per 40MHz SHARC - there's > just no way > that those DSPs had enough horsepower to do echo cancellation across > that manychannels. > > > This page http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html > seems to support what you say. It also implies DTMF detection sucks > unless you echo cancel. The statement "If the outgoing signal is a tone > of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" > is telling you to keep your outgoing signal in the same frequency range > as dial-tone where the dial-tone filter on the DTMF receiver will > obviate the need for an echo canceller. They are freely admitting > exactly what I have said. If you want a normal IVR with cut-through to > work you better turn that echo canceller on. > > My only experience with Aculab was fitting a box designed by other > people into a system. That one definitely echo cancelled, as it worked > as well as the Dialogic based boxes we developed ourselves. > > > That only holds true if your premise - that you need echo cancellation for > good > DTMF detection - is correct, which I don't believe it is. > > An Asterisk box with an el-cheapo quad E1 card in that I use for > TDM-SIP gatewaying > detects DTMF perfectly well with no echo cancellation. > > > You must have very low standards for "works well". > > > Nothing like a good old ad hominem attack. Beats reasoned argument any > day. > > You just don't need echo cancellation to achieve perfectly acceptable > DTMF detection. > > > Well, not if you expect people to wait for silence before entering DTMF, > but who would tolerate that these days? Cut through has been de rigeur > since the late 80s. > > > Oh, for pity's sake, you get perfectly good cut through without echo > cancellation. > Humour me and draw a quick mental picture of the spectrum of a random bit > of > speech at -20dBm; now add tones at -10dBm and -7dBm. They stick out like > a pair of sore thumbs. > > I'm sure it's quite possible to come up with a pathological case - e.g. > cut-through > against a 1kHz milliwatt tone, but that sort of thing just doesn't happen > in real- > life IVR applications. > > ASR - yes, maybe, but surely only in the case where the application > requires barge-in; > even then, I'd be interested to see some test results, particuarly > where the outbound prompt > is killed the moment the ASR reports start of speech. > > > Doesn't any sane system expect barge in to be nearly as reliable as > waiting for silence? Who would tolerate something that doesn't? It has > been a standard expectation of any decent IVR since they began. > > > Sorry - ASR with barge-in has been a standard expectation since the first > IVRs? > > I'm afraid that your original bald claim - that "IVRs badly need echo > cancellation" is simply > wrong, misleading and irresponsible: those believing it will end up > spending large sums > of money on technology which they probably do not need. > > > You must have very low standards for what works well. If you suggest > people leave out echo cancellation you are just asking for customer > service issues down the line. That whole Aculab page is a clear response > to just such issues they had, which forced them to add the necessary > improvements. > > > Repeating you ad-hominem really doesn't make it any stronger, I'm afraid. > And > the Aculab page you refer to offers four solutions for problems caused by > far- > end echo, of which cancellation is just one; not playing a stationary tone > above 600Hz > is another. > > Do you have any real-world samples of DTMF+echo which give your DTMF > detection code trouble? > > --Dave > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Arnaldo M Pereira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/153c92ec/attachment.html From mattdfong at gmail.com Tue Mar 17 21:39:36 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 18 Mar 2009 11:39:36 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> Message-ID: <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 2009/3/17 Anthony Minessale > there is a patch in jira that will implement this feature about to be added > > > > 2009/3/17 Matthew Fong > >> I apologize if this is a double post to -dev. I'm not sure why I don't see >> my message appearing, so I'm going to try again in the -user list (first >> timer posting here ;). >> >> I have a situation where it would be useful for a caller not to be hungup, >> after finishing the "fifo in" execution (when the agent disconnects the call >> or the agent hangs-up). The caller is automatically hungup, in this >> situation. It would be preferable if the caller channel went further along >> the dial plan. I thought I might get lucky implementing this setting with >> hangup_after_bridge to false, but fifo does not utilize this variable. >> I tried looking thru the mod_fifo.c source, but my c skills are minimal. I >> also tried executing fifo in a lua app and setting setAutoHangup(false), but >> that also did not work. Any chance this could be done as a feature >> enhancement? Thanks. >> >> --matt >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/91727d52/attachment-0001.html From mszlazak at aol.com Tue Mar 17 23:59:54 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 02:59:54 -0400 Subject: [Freeswitch-users] Is mod vmd working? Message-ID: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> I followed these instructions for Mod_vmd except for a Windows box: http://wiki.freeswitch.org/wiki/Mod_vmd I tried testing to see if it's working by dialing the following extension: ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ???? ??? ??? ??? ??? ??? ??? However, I didn't see channel variable "vmd_detect" in the FreeSwitch console. ?? Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/a83f0caf/attachment.html From sicfslist at gmail.com Wed Mar 18 00:11:49 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 18 Mar 2009 02:11:49 -0500 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> Message-ID: <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> Mark, It does work ... but I can't really attest to how well ... especially compared to other things out there. I started capturing this in CDR's to see and it didn't seem like it worked very well. If this is really critical to you, you might want to ping Ken Rice. I know he might have a better option. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/ac6546fd/attachment.html From codecomplete at free.fr Tue Mar 17 02:20:50 2009 From: codecomplete at free.fr (Gilles) Date: Tue, 17 Mar 2009 10:20:50 +0100 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? Message-ID: <7.0.1.0.2.20090317101702.02701c88@fredshack.com> Hello For single-host settings, getting customers to buy a separate server just to run Freeswitch is overkill, so I'm thinking about selling just the IVR application to run on Windows. Unless a PCI card is available, the FXO connection will be provided by Sangoma's USB device. I'd like some feedback on running Freeswitch on XP and Vista: Is it ready for production use? Does it require beefy hardware? Thank you for any hint. From kawarod at laposte.net Wed Mar 18 01:27:56 2009 From: kawarod at laposte.net (rod) Date: Wed, 18 Mar 2009 12:27:56 +0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <85B32876-7E49-4DDF-B92F-353DA9599DE9@avgs.ca> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> <49BA6608.70208@gmail.com> <49BA844C.3010409@freeswitch.org> <49BF4BF5.4080608@laposte.net> <85B32876-7E49-4DDF-B92F-353DA9599DE9@avgs.ca> Message-ID: <49C0B08C.4050008@laposte.net> thanks Mathieu. I setup an IRC account to give it a try. Comme ?a je pourrais t'embeter avec mes "pbms" :p rod Mathieu Rene wrote: > limit_hash uses a faster data structure then limit but works the same > way for tne end-user. > > viens sur IRC si t'as des questions en francais =) > > Math > > On 17-Mar-09, at 3:06 AM, rod wrote: > > >> Hi, >> >> not too hard :p >> but it's just a bad habit when I write in my native language >> (french). I >> guess that this spelling is not too common for english speaker. >> >> I'll do my best next time to write it correctly. >> >> @tamas >> you are right, we could use limit_hash the same way as limit when not >> specifying the /rate >> >> @Mathieu >> did you suggest limit_hash is more scalable than limit? But I don't >> understand why limit_hash is not suitable for data replication (DB >> lookup for limit and memory for limit_hash??), even if I don't know >> how >> to do it with limit. >> >> regards. >> >> Raymond Chandler wrote: >> >>> Tamas wrote: >>> >>> >>>> My guess is: pbm = problem :) >>>> >>>> >>>> >>> sure, but is it really that hard to spell all the way out? >>> >>> -Ray >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From oseslija at gmail.com Wed Mar 18 01:55:15 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Wed, 18 Mar 2009 09:55:15 +0100 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C03067.7070406@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> Message-ID: <4468a6770903180155r2fe4e1e1l91facb3085860982@mail.gmail.com> To share my experience: I had issues with echo with many E1 trunks in Serbia, especially when voice in between telco's network went to well known bad analog lines. I used OSLEC and I was fortunate to have Steve to complain to, he helped patching it further after my beta testing. Not many people would do that imho. I now switched to Sangoma cards with Octasic chips and occasionally would still hear certain echo. My view is that here some echo cancelling solution is very necessary, otherwise whole VoIP business comes up to bad reputation People would just not listen to themselves speaking, even using $400 phone. Regards, Ognjen 2009/3/18 David Knell > Steve Underwood wrote: > > [whopping big snip] > > > The first bit of that's a tad patronising, isn't it, > > > You are the one who started out being offensive. > > > I'm sorry if you find disagreement offensive; you might not wish to read > beyond this > point if so. > > and, in the case of the decade-old Aculab > cards which which I'm most familiar, is also untrue. > > > I can't find too much about the old cards on the web now, but I found http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html > which is pretty much a copy and paste from the old Dialogic web pages, > and you'll see it says "Cut through : Local echo cancellation permits > 100% detection with a >4.5 dB return loss line". The Aculabs did the > same thing for sure. They just couldn't work without cancellation. There > were some very early Dialogic cards, using DTMF receiver chips and OKI > ADPCM chips, and had no general purpose DSPs. They performed really > badly because of the lack of cancellation, and were quickly replaced > with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms > into a Motorola 56k DSP chips. > > > The same document, under the bit which you've quoted, says: > "(E-1) Digital trunks use separate transmit and receive paths to network. > Performance dependent on far end handset's match to local analog loop." > - i.e. the card does no echo cancellation. > > Aculab didn't even offer echo cancellation on Prosody for years and, when > they did, it > consumed prodigious amounts of DSP. Nonetheless, the DTMF detection worked > perfectly well, even across 120 channels per 40MHz SHARC - there's just no > way > that those DSPs had enough horsepower to do echo cancellation across that > many > channels. > > An Asterisk box with an el-cheapo quad E1 card in that I use for TDM-SIP > gatewaying > detects DTMF perfectly well with no echo cancellation. > > You just don't need echo cancellation to achieve perfectly acceptable DTMF > detection. > > ASR - yes, maybe, but surely only in the case where the application > requires barge-in; > even then, I'd be interested to see some test results, particuarly where > the outbound prompt > is killed the moment the ASR reports start of speech. > > I'm afraid that your original bald claim - that "IVRs badly need echo > cancellation" is simply > wrong, misleading and irresponsible: those believing it will end up > spending large sums > of money on technology which they probably do not need. > > --Dave > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/553ae739/attachment.html From helmut.kuper at ewetel.de Wed Mar 18 02:20:09 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 18 Mar 2009 10:20:09 +0100 Subject: [Freeswitch-users] openZAP disconnect cause wrong? Message-ID: <49C0BCC9.4000509@ewetel.de> Hello, I'm not sure whether the following is a bug or a config issue: I found this in my log file: 2009-03-18 10:07:00 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: USER_BUSY 2009-03-18 10:07:00 [DEBUG] mod_dptools.c:2025 audio_bridge_function() Continue on fail [true]: Cause: USER_BUSY 2009-03-18 10:07:00 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:5/2850 [CS_EXECUTE] [NORMAL_CLEARING] FS obviously doesn't pass through the disconnect cause from Bridge app to openzap module. Analyzing the corresponding q931.pcap trace confirms this. Do I have to configure it somewhere e.g. a mapping or so, or is this a bug? regrads helmut From helmut.kuper at ewetel.de Wed Mar 18 02:40:22 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 18 Mar 2009 10:40:22 +0100 Subject: [Freeswitch-users] openZAP disconnect cause wrong? In-Reply-To: <49C0BCC9.4000509@ewetel.de> References: <49C0BCC9.4000509@ewetel.de> Message-ID: <49C0C186.3070307@ewetel.de> Hello, ok found it ... was a configuration issue due to the "continue on fail = true" variable in my dialplan. "Hangup" application fixed this :) Sorry for the post. regards helmut On 18.03.2009 10:20, Helmut Kuper wrote: > Hello, > > I'm not sure whether the following is a bug or a config issue: > > From andy at fabulous4.co.uk Wed Mar 18 04:20:11 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Wed, 18 Mar 2009 11:20:11 -0000 Subject: [Freeswitch-users] Losing Gateway registration Message-ID: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/6df3c513/attachment-0001.html From mszlazak at aol.com Wed Mar 18 05:04:37 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 08:04:37 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> Message-ID: <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> SDR? I'm wondering why there was nothing in the console showing the channel variable ${vmd_detect} as the wiki says there should be: Mark -----Original Message----- From: Shelby Ramsey To: freeswitch-users at lists.freeswitch.org Sent: Wed, 18 Mar 2009 12:11 am Subject: Re: [Freeswitch-users] Is mod vmd working? Mark, It does work ... but I can't really attest to how well ... especially compared to other things out there.? I started capturing this in CDR's to see and it didn't seem like it worked very well. If this is really critical to you, you might want to ping Ken Rice.? I know he might have a better option. SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/17cb612a/attachment.html From dave at 3c.co.uk Wed Mar 18 06:00:12 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 18 Mar 2009 07:00:12 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> Message-ID: <49C0F05C.5090204@3c.co.uk> Hi Arnaldo, That's interesting - Brasil was my first proper IVR installation: one with Embratel in Sao Paulo, and then a couple with TeleRJ. I remember landing at Sao Paulo airport for the first time at 7 a.m. with instructions to "meet a fat man called Ferrari" unsure as to whether I was in some sort of elaborate hoax (I wasn't, and he was), and learning my first three words of Portuguese as we left the car park: filho da puta, of course. Those had no EC. DTMF detection worked fine, and the audio quality of the IVR recordings was perfect, which is what you'd expect: EC doesn't alter the IVR->caller audio at all. A TDM->SIP->TDM type application is a different animal: you've got the added latency of packetisation/jitter buffering/etc. which pretty much makes echo cancellation a must. --Dave > Sharing my humble experience: in Brazil we usually need echo > cancellation to have reliable DTMF detection _and_ voice quality over > E1 lines (be it on MFC/R2 - r2d - or ISDN PRI lines), either for > sip/tdm gateway devices or IVR applications. > > Usually there's no need for echo cancellation on links from some > Telcos, in some specific places. But we need it in the majority of > cases, even when my box is just a gateway between legacy pbxes. > > This represents just a subset of the available E1s in the world and > it's just a practical experience, but it's a fact for me. If I don't > have a card with echo cancellation, I don't offer reliability to my > customer; I've done that in the past and didn't work out. > > I'm not theoretically discussing anything, just sharing what I've been > through in the last 4 or 5 years. From sicfslist at gmail.com Wed Mar 18 06:07:55 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 18 Mar 2009 08:07:55 -0500 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> Message-ID: <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> Mark, Because it didn't detect a "beep". It will be be there as vmd_detect=true if it does. I'm not sure exactly how reliable it's "beep" detection is. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/36bc0a0c/attachment.html From steveu at coppice.org Wed Mar 18 06:12:10 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 21:12:10 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C05641.7070309@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> Message-ID: <49C0F32A.90802@coppice.org> OK, one last go and I give up. Lets look at the documentation for Dialogic springware. This is the DSP package that loads in their cards or runs on the host in HMP applications. It does things like DTMF generation and detection for all Dialogic cards except the DM3 series. The documentation says: *PerfectDigit DTMF Signaling* ? DSP-based DTMF (touchtone) detection algorithm optimized for lowest talk-off and play-off susceptibility in the industry. The system will not easily be fooled by mistaking human speech for DTMF tones. ? Minimum tone duration and interdigit delay times accurately handle speed dialing typical of "power users" ? Utilizes echo cancellation which results in superior cut through for accurate DTMF tone interpretation during voice file playback within a broad range of network/switch environments ? DTMF outbound dialing generated by DSP for accuracy and flexibility (dialing levels are adjustable to meet a variety of global PTT requirements) Detecting supervisory tones on phone lines is trivial. Not falsely detecting them is where things get interesting. The standard test for DTMF receivers is a set of cassette tapes from Bellcore containing about 3 hours of snippets from real telephone calls in North America. Most DTMF receiver chips get a few hundred false DTMF hits in those 3 hours. Dialogic get 20 something. My DTMF receiver gets 19. The reason its hard to detect these things reliably is voice doesn't sit there nicely at one level. Its level and its spectrum bounce all over the place, and a real DTMF digit is only there for 40ms or so. I defy anyone to visually identify a 40ms DTMF digit amongst real dynamic speech if it isn't *way* above the voice in amplitude. This is why your phone has to mute your voice when you press a digit. The DTMF receiver has no chance of reliable detection with speech and digits mixed. In the few special cases where concurrent speech and signaling tone are present on the PSTN (e.g. 2280Hz signaling in .eu and 2600Hz in .us) the signaling sequence is very carefully constructed to avoid confusing the system. DTMF is never used in that way. There is one obvious special case where all DTMF receivers need to tolerate spillback. They need to differentiate between dialing tone and DTMF on the first digit you dial. They do this very simply. Dialing tone was chosen to be pretty low frequencies - 350Hz + 440Hz, 425Hz + 475Hz and similar pairings. The lowest DTMF tone is well above this. An aggressive low pass filter in the DTMF receiver removes the dial tone spillback, while barely affecting the lowest DTMF tone. This was the original design of DTMF, but...... IVRs changed all that. Their DTMF receivers are expected to work amidst outgoing prompts, which may be going to phones with an awful match to the line. The spillback can be huge. The good IVR hardware suppliers, like Dialogic, very quickly added echo cancellation to their cards. I can say a lot of negative things about Dialogic, but one thing they did really well was their DTMF cut-through. When people get used to an IVR they expect to hammer in digit sequences as fast as they can, in the face of a machine desperately trying to play voice prompts to them. Dialogic cards do this really well, on lines of all types, and on networks of varying quality. This would be impossible without echo cancellation. David Knell wrote: > Steve Underwood wrote: >> David Knell wrote: >> >>> Steve Underwood wrote: >>> >>>> [whopping big snip] >>>> >>>> >>>>> The first bit of that's a tad patronising, isn't it, >>>>> >>>>> >>>> You are the one who started out being offensive. >>>> >>>> >>> I'm sorry if you find disagreement offensive; you might not wish to >>> read beyond this >>> point if so. >>> >>>>> and, in the case of the decade-old Aculab >>>>> cards which which I'm most familiar, is also untrue. >>>>> >>>>> >>>> I can't find too much about the old cards on the web now, but I found >>>> http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html >>>> which is pretty much a copy and paste from the old Dialogic web pages, >>>> and you'll see it says "Cut through : Local echo cancellation permits >>>> 100% detection with a >4.5 dB return loss line". The Aculabs did the >>>> same thing for sure. They just couldn't work without cancellation. There >>>> were some very early Dialogic cards, using DTMF receiver chips and OKI >>>> ADPCM chips, and had no general purpose DSPs. They performed really >>>> badly because of the lack of cancellation, and were quickly replaced >>>> with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms >>>> into a Motorola 56k DSP chips. >>>> >>>> >>> The same document, under the bit which you've quoted, says: >>> "(E-1) Digital trunks use separate transmit and receive paths to network. >>> Performance dependent on far end handset's match to local analog loop." >>> - i.e. the card does no echo cancellation. >>> >> Your messages are starting to looked deranged. Why would they only apply >> echo cancellation to T1s? Its a bizarre idea, and you must realise its >> wrong. Are you so desperate to support a wrong answer you'll clutch at >> straws? :-\ >> > More insults. Answer me this: if there were echo cancellation in use, > why would > DTMF detection performance depend on the far-end handset's match to > the loop? > > And the follow-up question (which you've already pretty much asked) - > if the > card doesn't echo cancel for E1s, why would it for T1s? > > As an aside, I'm not convinced that the document's not talking about > return loss > on the T1 line itself, the implication being that the T1 is being > carried on a single > pair, which makes the first sentence about E1s make a bit more sense. > But that's > just a guess. >>> Aculab didn't even offer echo cancellation on Prosody for years and, >>> when they did, it >>> consumed prodigious amounts of DSP. Nonetheless, the DTMF detection >>> worked >>> perfectly well, even across 120 channels per 40MHz SHARC - there's >>> just no way >>> that those DSPs had enough horsepower to do echo cancellation across >>> that manychannels. >>> >> This page >> http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html >> seems to support what you say. It also implies DTMF detection sucks >> unless you echo cancel. The statement "If the outgoing signal is a tone >> of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" >> is telling you to keep your outgoing signal in the same frequency range >> as dial-tone where the dial-tone filter on the DTMF receiver will >> obviate the need for an echo canceller. They are freely admitting >> exactly what I have said. If you want a normal IVR with cut-through to >> work you better turn that echo canceller on. >> >> My only experience with Aculab was fitting a box designed by other >> people into a system. That one definitely echo cancelled, as it worked >> as well as the Dialogic based boxes we developed ourselves. >> > That only holds true if your premise - that you need echo cancellation > for good > DTMF detection - is correct, which I don't believe it is. >>> An Asterisk box with an el-cheapo quad E1 card in that I use for >>> TDM-SIP gatewaying >>> detects DTMF perfectly well with no echo cancellation. >>> >> You must have very low standards for "works well". >> > Nothing like a good old ad hominem attack. Beats reasoned argument any > day. >>> You just don't need echo cancellation to achieve perfectly acceptable >>> DTMF detection. >>> >> Well, not if you expect people to wait for silence before entering DTMF, >> but who would tolerate that these days? Cut through has been de rigeur >> since the late 80s. >> > Oh, for pity's sake, you get perfectly good cut through without echo > cancellation. > Humour me and draw a quick mental picture of the spectrum of a random > bit of > speech at -20dBm; now add tones at -10dBm and -7dBm. They stick out like > a pair of sore thumbs. > > I'm sure it's quite possible to come up with a pathological case - > e.g. cut-through > against a 1kHz milliwatt tone, but that sort of thing just doesn't > happen in real- > life IVR applications. >>> ASR - yes, maybe, but surely only in the case where the application >>> requires barge-in; >>> even then, I'd be interested to see some test results, particuarly >>> where the outbound prompt >>> is killed the moment the ASR reports start of speech. >>> >> Doesn't any sane system expect barge in to be nearly as reliable as >> waiting for silence? Who would tolerate something that doesn't? It has >> been a standard expectation of any decent IVR since they began. >> > Sorry - ASR with barge-in has been a standard expectation since the > first IVRs? >>> I'm afraid that your original bald claim - that "IVRs badly need echo >>> cancellation" is simply >>> wrong, misleading and irresponsible: those believing it will end up >>> spending large sums >>> of money on technology which they probably do not need. >>> >> You must have very low standards for what works well. If you suggest >> people leave out echo cancellation you are just asking for customer >> service issues down the line. That whole Aculab page is a clear response >> to just such issues they had, which forced them to add the necessary >> improvements. >> > Repeating you ad-hominem really doesn't make it any stronger, I'm > afraid. And > the Aculab page you refer to offers four solutions for problems caused > by far- > end echo, of which cancellation is just one; not playing a stationary > tone above 600Hz > is another. Doesn't "don't use frequencies above 600Hz" mean they won't work very well with voice present? Their "solutions" are use echo cancellation or don't create cut-through situations. That whole Aculab page is skirting around making a direct statement that reliable cut-through demands echo cancellation. > > Do you have any real-world samples of DTMF+echo which give your DTMF > detection code trouble? Any analogue line with substantial spillback. Huge numbers of lines have spillback gain only a few dB below the true receive gain. You don't normally notice this, as with a short delay its just a bit of pleasant reverb. Its becomes a problem with high latency VoIP paths stretch that delay. Its also a problem for any equipment which needs a clean signal from the far end. A modem for example, or an IVR. Echo cancellation is the only practical solution. Regards, Steve From nik.middleton at noblesolutions.co.uk Wed Mar 18 06:16:46 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Mar 2009 13:16:46 -0000 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> Message-ID: Another issue with this module is the resources it consumes. We had it running on 50 calls yesterday and the cpu's all went to 90+% Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shelby Ramsey Sent: 18 March 2009 13:08 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Is mod vmd working? Mark, Because it didn't detect a "beep". It will be be there as vmd_detect=true if it does. I'm not sure exactly how reliable it's "beep" detection is. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/4272f8e3/attachment-0001.html From steveu at coppice.org Wed Mar 18 06:25:44 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 21:25:44 +0800 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> Message-ID: <49C0F658.4020006@coppice.org> Nik Middleton wrote: > > Another issue with this module is the resources it consumes. We had it > running on 50 calls yesterday and the cpu?s all went to 90+% > That's odd. Something must be fouling up, as the algorithm he used should be very lightweight. Steve From intralanman at freeswitch.org Wed Mar 18 06:40:58 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 18 Mar 2009 09:40:58 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C0F05C.5090204@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> Message-ID: <49C0F9EA.3000200@freeswitch.org> David Knell wrote: > Hi Arnaldo, > > That's interesting - Brasil was my first proper IVR installation: one > with Embratel in Sao Paulo, and then a couple with TeleRJ. I remember > landing at Sao Paulo airport for the first time at 7 a.m. with > instructions to "meet a fat man called Ferrari" unsure as to whether I > was in some sort of elaborate hoax (I wasn't, and he was), and learning > my first three words of Portuguese as we left the car park: filho da > puta, of course. > What's interesting to me is.... everyone on this thread except you has said that in real-world scenarios, they need the EC for reliability. One of which, does signal processing programming professionally. It seems to me that if you "build a better mouse trap" you must know what's involved in making it work properly. I'm not sure what your background really is, but you'd be hard pressed to match up to Steve's reputation and/or experience. That said, it might be a good idea to just agree to disagree as this is starting to sound like the faxing over IP talks I hear a lot. (i.e. "faxing over g.711u with no t.38 works fine for me") Where it might work for some people by some mysterious phenomena, it won't work for the general public. And telling people that they don't need EC, where so many have already said that they obviously do, is just as irresponsible, IMHO, as you claiming Steve was for telling them that they don't need it. -Ray From brian at freeswitch.org Wed Mar 18 07:07:55 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Mar 2009 09:07:55 -0500 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: References: Message-ID: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: > Hi, > > I've recently ugrade to version 1.02 of freeswitch and am having > some problems with my gateway registrations. The gateway > successfully registers with my voip provider when freeswitch first > starts but if left running it seems to loose it's connection to my > voip provider. I can get it to reconnect with a sofia restart. I'm > using the same provider and user account as with the old version of > the software. Can you suggest any reaosn why this may be happening > and how I can prevent it? > > Many thanks > Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/ec96a7e9/attachment.html From mrene_lists at avgs.ca Wed Mar 18 07:45:39 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 18 Mar 2009 10:45:39 -0400 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> References: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> Message-ID: <2365564C-92A8-483E-9FCB-34EBE71EC256@avgs.ca> if you are behind NAT it is possible that your router "forgot" the mapping betweeen FS and your provider, try adding to your gateway. Math On 18-Mar-09, at 10:07 AM, Brian West wrote: > Upgrade to 1.03 or SVN Trunk > > /b > > On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: > >> Hi, >> >> I've recently ugrade to version 1.02 of freeswitch and am having >> some problems with my gateway registrations. The gateway >> successfully registers with my voip provider when freeswitch first >> starts but if left running it seems to loose it's connection to my >> voip provider. I can get it to reconnect with a sofia restart. I'm >> using the same provider and user account as with the old version of >> the software. Can you suggest any reaosn why this may be happening >> and how I can prevent it? >> >> Many thanks >> Andy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/ae829584/attachment.html From mszlazak at aol.com Wed Mar 18 08:32:32 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 11:32:32 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> Message-ID: <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> I added a voicemail tag in 5555 to a default extension 1001, I hear the voicemail beep but still don't see vmd_detect. Mark -----Original Message----- From: Shelby Ramsey To: freeswitch-users at lists.freeswitch.org Sent: Wed, 18 Mar 2009 6:07 am Subject: Re: [Freeswitch-users] Is mod vmd working? Mark, Because it didn't detect a "beep".? It will be be there as vmd_detect=true if it does.? I'm not sure exactly how reliable it's "beep" detection is.? SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/b34a1129/attachment.html From mike at jerris.com Wed Mar 18 08:33:28 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Mar 2009 11:33:28 -0400 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: <7.0.1.0.2.20090317101702.02701c88@fredshack.com> References: <7.0.1.0.2.20090317101702.02701c88@fredshack.com> Message-ID: There is currently no openzap (sangoma, etc) support on windows, we hope this will be coming soon. Mike On Mar 17, 2009, at 5:20 AM, Gilles wrote: > Hello > > For single-host settings, getting customers to buy a separate server > just to run Freeswitch is overkill, so I'm thinking about selling > just the IVR application to run on Windows. Unless a PCI card is > available, the FXO connection will be provided by Sangoma's USB > device. > > I'd like some feedback on running Freeswitch on XP and Vista: Is it > ready for production use? Does it require beefy hardware? > > Thank you for any hint. From mike at jerris.com Wed Mar 18 08:34:24 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Mar 2009 11:34:24 -0400 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: <20090318023154.GA16523@jdc.jasonjgw.net> References: <20090318014659.GA15840@jdc.jasonjgw.net> <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> <20090318023154.GA16523@jdc.jasonjgw.net> Message-ID: On Mar 17, 2009, at 10:31 PM, Jason White wrote: > Brian West wrote: >> if you installed the ssl devel stuff AFTER you configured you'll need >> to reconfigure. > > I'm reasonably sure it was installed already, unless it was pulled > in recently > by a package upgrade. > > The configure script needs to look in /usr/include/openssl for the > headers. > I'll have a look at config.log and try to work out what it looked > for and why > it didn't find it. you will have to look in the config.log in libs/sofia-sip Mike From Mark.Tabron at rnid-typetalk.org.uk Wed Mar 18 08:52:14 2009 From: Mark.Tabron at rnid-typetalk.org.uk (MarkTab) Date: Wed, 18 Mar 2009 08:52:14 -0700 (PDT) Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> Message-ID: <22582281.post@talk.nabble.com> We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things! Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819 http://pastebin.freeswitch.org/7819 I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818 http://pastebin.freeswitch.org/7818 Getting there (slowly) Mark. mercutioviz wrote: > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: >> Another update - this time (part) good news! Decided to run wancfg_tdmapi >> again, using the same settings as we always did, and we can now make >> external calls. I suspect that whatever BT did yesterday kicked the >> circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > >> >> However placing an external call into FS isn't as successful, looks like >> it can't assign a channel and terminates the call. >> > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Problem-dialing-out-via-E1-tp22479047p22582281.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Mark.Tabron at rnid-typetalk.org.uk Wed Mar 18 08:55:50 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Wed, 18 Mar 2009 15:55:50 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> Message-ID: <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things! Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819 I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818 Getting there (slowly) Mark. On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron wrote: > Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed." > > However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. > Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 March 2009 15:48 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron wrote: > Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed." > > However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. > Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- From mattdfong at gmail.com Wed Mar 18 08:59:07 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 18 Mar 2009 22:59:07 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> Message-ID: <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> I upgraded to FreeSWITCH Version 1.0.trunk (12654M) but caller is still being hungup (and not continuing on with dialplan) after agent disconnect with hangup_after_bridge=false Is there a separate patch I need to apply? Thanks. --matt On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: > Hi Anthony, thanks for the reply. > I've searched thru jira, and didn't find anything when searching for fifo > that was recently updated or related, except > > http://jira.freeswitch.org/browse/MODAPP-189 > > and I'm not sure if this does what I need. Was this what you were referring > to? Thanks. > > --matt > > 2009/3/17 Anthony Minessale > > there is a patch in jira that will implement this feature about to be added >> >> >> >> 2009/3/17 Matthew Fong >> >>> I apologize if this is a double post to -dev. I'm not sure why I don't >>> see my message appearing, so I'm going to try again in the -user list (first >>> timer posting here ;). >>> >>> I have a situation where it would be useful for a caller not to be >>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>> the call or the agent hangs-up). The caller is automatically hungup, in this >>> situation. It would be preferable if the caller channel went further along >>> the dial plan. I thought I might get lucky implementing this setting with >>> hangup_after_bridge to false, but fifo does not utilize this variable. >>> I tried looking thru the mod_fifo.c source, but my c skills are minimal. >>> I also tried executing fifo in a lua app and setting setAutoHangup(false), >>> but that also did not work. Any chance this could be done as a feature >>> enhancement? Thanks. >>> >>> --matt >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/9bec90cd/attachment.html From Prometheus001 at gmx.net Wed Mar 18 10:18:16 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 18 Mar 2009 18:18:16 +0100 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <49C12CD8.7020203@gmx.net> 2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch there was a timer problem which was not solved yet. This caused channels to be busy in my case. I am not sure whether this is solved yet. Can anybody confirm? Best regards Peter Mark Tabron schrieb: > We're a couple more steps forward from yesterday. Turned out some of my > regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has > an extra space before one of the closing brackets in the default.xml > example. After staring at the screen all day it's funny how you miss > these things! > > Situation now is I can get the call into FS but, it rings the extension > for a fraction of a second then the call drops. Here's the contents of > the public and default dialplans I'm using (as per example in the wiki) > and the debug - http://pastebin.freeswitch.org/7819 > > I'm also seeing another issue when placing subsequent inbound calls, > they bounce if hitting the same channel the first call came in to > (typically /1:1). Again, grabbed a debug of this - > http://pastebin.freeswitch.org/7818 > > Getting there (slowly) > > Mark. > > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: > >> Another update - this time (part) good news! Decided to run >> > wancfg_tdmapi again, using the same settings as we always did, and we > can now make external calls. I suspect that whatever BT did yesterday > kicked the circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > > >> However placing an external call into FS isn't as successful, looks >> > like it can't assign a channel and terminates the call. > > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 17 March 2009 15:48 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: > >> Another update - this time (part) good news! Decided to run >> > wancfg_tdmapi again, using the same settings as we always did, and we > can now make external calls. I suspect that whatever BT did yesterday > kicked the circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > > >> However placing an external call into FS isn't as successful, looks >> > like it can't assign a channel and terminates the call. > > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > -------------------------------------------------------------------------------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > -------------------------------------------------------------------------------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Wed Mar 18 10:24:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Mar 2009 10:24:01 -0700 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> Message-ID: <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> 2009/3/18 : > I added a voicemail tag in 5555 to a default extension 1001, I hear the > voicemail beep but still don't see vmd_detect. > > Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering machine beeps and vm beeps on cell phone voice mails. -MC From msc at freeswitch.org Wed Mar 18 10:27:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Mar 2009 10:27:10 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <49C12CD8.7020203@gmx.net> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> <49C12CD8.7020203@gmx.net> Message-ID: <87f2f3b90903181027y5937f48bxa5aecea367db4cc0@mail.gmail.com> On Wed, Mar 18, 2009 at 10:18 AM, Peter P GMX wrote: > 2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch > there was a timer problem which was not solved yet. This caused channels > to be busy in my case. > > I am not sure whether this is solved yet. Can anybody confirm? > We're using ozmod_libpri which has it's own PRI handling. So far, so good... -MC From anthony.minessale at gmail.com Wed Mar 18 12:00:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Mar 2009 14:00:14 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> Message-ID: <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> This is the patch http://jira.freeswitch.org/browse/MODAPP-237 it's not added yet. 2009/3/18 Matthew Fong > I upgraded to > FreeSWITCH Version 1.0.trunk (12654M) > > but caller is still being hungup (and not continuing on with dialplan) > after agent disconnect with hangup_after_bridge=false > > Is there a separate patch I need to apply? Thanks. > > --matt > > > On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: > >> Hi Anthony, thanks for the reply. >> I've searched thru jira, and didn't find anything when searching for fifo >> that was recently updated or related, except >> >> http://jira.freeswitch.org/browse/MODAPP-189 >> >> and I'm not sure if this does what I need. Was this what you were >> referring to? Thanks. >> >> --matt >> >> 2009/3/17 Anthony Minessale >> >> there is a patch in jira that will implement this feature about to be >>> added >>> >>> >>> 2009/3/17 Matthew Fong >>> >>>> I apologize if this is a double post to -dev. I'm not sure why I don't >>>> see my message appearing, so I'm going to try again in the -user list (first >>>> timer posting here ;). >>>> >>>> I have a situation where it would be useful for a caller not to be >>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>> situation. It would be preferable if the caller channel went further along >>>> the dial plan. I thought I might get lucky implementing this setting with >>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>> I tried looking thru the mod_fifo.c source, but my c skills are minimal. >>>> I also tried executing fifo in a lua app and setting setAutoHangup(false), >>>> but that also did not work. Any chance this could be done as a feature >>>> enhancement? Thanks. >>>> >>>> --matt >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/0d7c7233/attachment-0001.html From mszlazak at aol.com Wed Mar 18 15:12:20 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 18:12:20 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com><35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com><8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> Message-ID: <8CB762DBDEB5479-1554-817@mblk-d43.sysops.aol.com> Hi MC, With trunk 12638M, I tried checking vmd internally and externally to my cell. No luck at all in detecting a voicemail (beep). I used the following extensions to test this, maybe they are in error. If not then how else can I detect from FS that I got voicemail in a phone agnostic way (i.e, pots & sip). ??? ??? ??? Mark. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 18 Mar 2009 10:24 am Subject: Re: [Freeswitch-users] Is mod vmd working? 2009/3/18 : > I added a voicemail tag in 5555 to a default extension 1001, I hear the > voicemail beep but still don't see vmd_detect. > > Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering machine beeps and vm beeps on cell phone voice mails. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/3c414272/attachment.html From kjv at ken-ton.com Wed Mar 18 15:39:59 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Wed, 18 Mar 2009 18:39:59 -0400 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: References: <20090318014659.GA15840@jdc.jasonjgw.net> <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> <20090318023154.GA16523@jdc.jasonjgw.net> Message-ID: Was this ever resolved? If we're missing something in the documentation, I'd like to make sure it's in there. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3239 x0 On Mar 18, 2009, at 11:34 AM, Michael Jerris wrote: > > On Mar 17, 2009, at 10:31 PM, Jason White wrote: > >> Brian West wrote: >>> if you installed the ssl devel stuff AFTER you configured you'll >>> need >>> to reconfigure. >> >> I'm reasonably sure it was installed already, unless it was pulled >> in recently >> by a package upgrade. >> >> The configure script needs to look in /usr/include/openssl for the >> headers. >> I'll have a look at config.log and try to work out what it looked >> for and why >> it didn't find it. > > you will have to look in the config.log in libs/sofia-sip > > Mike > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/4a424864/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/4a424864/attachment.bin From brian at freeswitch.org Wed Mar 18 15:43:40 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Mar 2009 17:43:40 -0500 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: References: <20090318014659.GA15840@jdc.jasonjgw.net> <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> <20090318023154.GA16523@jdc.jasonjgw.net> Message-ID: I thought we had... hrm. /b On Mar 18, 2009, at 5:39 PM, Karl Vesterling wrote: > Was this ever resolved? > If we're missing something in the documentation, I'd like to make > sure it's in there. > > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3239 x0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/d72d9983/attachment.html From nik.middleton at noblesolutions.co.uk Wed Mar 18 16:22:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Mar 2009 23:22:14 -0000 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com><35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com><8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> Message-ID: Hmm, Well We're connected direct to E1's and it doesn't work reliably here. That said, DTMF detect does recognise the beeps most of the time. Perhaps there's a regional variation. I wonder if it's country specific. The code looks logical. When I get some time I'll have a look at it and see how it can be improved. The concept is great and is much better that sniffing out human voice as that's prone to false positives. Much better to assume human and machine. Nothing worse than a silent call. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 March 2009 17:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Is mod vmd working? 2009/3/18 : > I added a voicemail tag in 5555 to a default extension 1001, I hear the > voicemail beep but still don't see vmd_detect. > > Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering machine beeps and vm beeps on cell phone voice mails. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed Mar 18 17:15:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Mar 2009 17:15:34 -0700 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> Message-ID: <87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com> Ironically, I've used tone_detect to try and trap SIT tones and I found that answering machines in the USA seem to all send a beep in the same freq range as American SIT tones... :) -MC On Wed, Mar 18, 2009 at 4:22 PM, Nik Middleton wrote: > Hmm, > > Well We're connected direct to E1's and it doesn't work reliably here. > That said, DTMF detect does recognise the beeps most of the time. > Perhaps there's a regional variation. ?I wonder if it's country > specific. ?The code looks logical. ?When I get some time I'll have a > look at it and see how it can be improved. > > The concept is great and is much better that sniffing out human voice as > that's prone to false positives. ?Much better to assume human and > machine. ?Nothing worse than a silent call. > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 18 March 2009 17:24 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Is mod vmd working? > > 2009/3/18 ?: >> I added a voicemail tag in 5555 to a default extension 1001, I hear > the >> voicemail beep but still don't see vmd_detect. >> >> Mark > > FYI, I've used mod_vmd but only in a TDM environment on outbound calls > via a PRI. It worked very well on for detecting answering ?machine > beeps and vm beeps on cell phone voice mails. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Wed Mar 18 20:16:45 2009 From: dujinfang at gmail.com (dujinfang) Date: Thu, 19 Mar 2009 11:16:45 +0800 Subject: [Freeswitch-users] Is there a way to automatically re-login gtalk account Message-ID: Hi all, mod_dingaling in client mode works well for me, but disconnected yesterday. 2009-03-18 16:57:32 [DEBUG] libdingaling.c:1545 xmpp_connect() io error 2 7 I use dl_login profile=gmail.com, and it re-login successfully. Is their a way to auto re-login after fail? Thanks. From mszlazak at aol.com Wed Mar 18 20:20:18 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 23:20:18 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com><35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com><8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com><87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> <87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com> Message-ID: <8CB7658C3800442-84C-1EAB@WEBMAIL-DY38.sysops.aol.com> tone_detect! sounds good. BTW, was there any errors in those extensions I posted. I modified something you posted MC. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 18 Mar 2009 5:15 pm Subject: Re: [Freeswitch-users] Is mod vmd working? Ironically, I've used tone_detect to try and trap SIT tones and I found that answering machines in the USA seem to all send a beep in the same freq range as American SIT tones... :) -MC On Wed, Mar 18, 2009 at 4:22 PM, Nik Middleton wrote: > Hmm, > > Well We're connected direct to E1's and it doesn't work reliably here. > That said, DTMF detect does recognise the beeps most of the time. > Perhaps there's a regional variation. ?I wonder if it's country > specific. ?The code looks logical. ?When I get some time I'll have a > look at it and see how it can be improved. > > The concept is great and is much better that sniffing out human voice as > that's prone to false positives. ?Much better to assume human and > machine. ?Nothing worse than a silent call. > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 18 March 2009 17:24 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Is mod vmd working? > > 2009/3/18 ?: >> I added a voicemail tag in 5555 to a default extension 1001, I hear > the >> voicemail beep but still don't see vmd_detect. >> >> Mark > > FYI, I've used mod_vmd but only in a TDM environment on outbound calls > via a PRI. It worked very well on for detecting answering ?machine > beeps and vm beeps on cell phone voice mails. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/657e01ac/attachment.html From d at d-man.org Wed Mar 18 20:27:52 2009 From: d at d-man.org (Darren Schreiber) Date: Wed, 18 Mar 2009 20:27:52 -0700 Subject: [Freeswitch-users] Issue relating mod_nibblebill In-Reply-To: References: Message-ID: Is this issue still open? I just noticed this. The error you are receiving indicates UnixODBC is installed, but not configured properly (most likely anyway). The UnixODBC drivers are kind of a pain to setup on some systems, especially CentOS, but this article may help you get it working - http://webaj.com/how-setup-mysql-dsn-datasbase-source-centos-myodbc-and-unix odbc-command-line.htm. I strongly recommend making sure the test commands they list work before trying to get UnixODBC working within FreeSWITCH. Also, it looks like you may have failed to copy the sample mod_nibblebill XML config file to /usr/local/freeswitch/conf/autoload_configs/ . You may want to give that a try. Within that file is the name of the ODBC driver being referenced - make sure that driver exists (see link above). - Darren -----Original Message----- From: JayaPrakash [mailto:jp.manchu at gmail.com] Sent: Saturday, March 14, 2009 4:49 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Issue relating mod_nibblebill Hi All, I am newbie to Freeswitch. I installed freeswitch-1.0.3 in Debian machine. I am able to make call, check presence, retrieve CDRs. I followed the installation steps given in mod_nibblebill for rating. While, installing mysql-connector-odbc, it has thrown errors related to mysql-config file, that it does not exist. Coming to mysql, mysql-client-5 and mysql-server-5 are installed. So I installed libmyodbc which is used for the same functionality. Rest of the steps are done, as given in mod-nibblebill installation. When the freeradius server is restarted, it has given the following error. 2009-03-14 14:55:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-03-14 14:55:47 [CRIT] mod_nibblebill.c:233 load_config() Cannot connect to ODBC driver/database freeswitch (user: root / pass dev)! Will you please have a look in solving this issue ? , how the issue can be solved? Thanks & Regards, JP. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dujinfang at gmail.com Wed Mar 18 20:30:23 2009 From: dujinfang at gmail.com (seven) Date: Thu, 19 Mar 2009 11:30:23 +0800 Subject: [Freeswitch-users] Is there a way to automatically re-login gtalk account Message-ID: <49963D01-BDB1-4142-8B1F-8F22773D950E@gmail.com> I do have auto-login enabled in jingle_profile: From pablosaro at gmail.com Wed Mar 18 21:24:46 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Thu, 19 Mar 2009 01:24:46 -0300 Subject: [Freeswitch-users] FS in Solaris Message-ID: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> Hi list, Any experience building FS in Solaris using Sun Studio? Thanks Pablo From switchserver at gmail.com Thu Mar 19 00:41:26 2009 From: switchserver at gmail.com (HarryK) Date: Thu, 19 Mar 2009 00:41:26 -0700 (PDT) Subject: [Freeswitch-users] Cepstral and RSS feeds Message-ID: <22594923.post@talk.nabble.com> I have Cepstral working. Can someone please tell me how to go about having it read RSS feeds? I can have the dialplan direct it np. But I really dont have a clue how to point it at an RSS. Any help would be great, ddint find anything in the wiki. -- View this message in context: http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22594923.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yudha2008 at gmail.com Thu Mar 19 00:53:50 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 19 Mar 2009 13:23:50 +0530 Subject: [Freeswitch-users] dynamic conference In-Reply-To: <21123756.post@talk.nabble.com> References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com> <21054080.post@talk.nabble.com> <49491847.3030403@lists.rupa.com> <21069519.post@talk.nabble.com> <494A9EE2.7050507@freeswitch.org> <21123756.post@talk.nabble.com> Message-ID: *Hi, I have seen the above mail. In that all of you tried to created dynamic conference through diaplan itself using the database to insert the uuid, caller_id_number, destination_number, etc .Can one guide me set the dynamic conference and Schema for the dynamic conference. I have tried the above dialplan with my Freeswitch 1.0.2 server but i cant able to get it. Can any one guide me for the this process. -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/19c81283/attachment.html From Mark.Tabron at rnid-typetalk.org.uk Thu Mar 19 02:08:28 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Thu, 19 Mar 2009 09:08:28 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> <49C12CD8.7020203@gmx.net> Message-ID: <11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL> So the second issue is possibly known - really could do with a fix or a workaround for this as we plan to use E1's for all incoming traffic. Can anyone shed light on the first problem (extension rings for a fraction of a second then hangs up) I mentioned below, or is that possibly part of the same issue? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P GMX Sent: 18 March 2009 17:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch there was a timer problem which was not solved yet. This caused channels to be busy in my case. I am not sure whether this is solved yet. Can anybody confirm? Best regards Peter Mark Tabron schrieb: > We're a couple more steps forward from yesterday. Turned out some of my > regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has > an extra space before one of the closing brackets in the default.xml > example. After staring at the screen all day it's funny how you miss > these things! > > Situation now is I can get the call into FS but, it rings the extension > for a fraction of a second then the call drops. Here's the contents of > the public and default dialplans I'm using (as per example in the wiki) > and the debug - http://pastebin.freeswitch.org/7819 > > I'm also seeing another issue when placing subsequent inbound calls, > they bounce if hitting the same channel the first call came in to > (typically /1:1). Again, grabbed a debug of this - > http://pastebin.freeswitch.org/7818 > > Getting there (slowly) > > Mark. > > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: > >> Another update - this time (part) good news! Decided to run >> > wancfg_tdmapi again, using the same settings as we always did, and we > can now make external calls. I suspect that whatever BT did yesterday > kicked the circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > > >> However placing an external call into FS isn't as successful, looks >> > like it can't assign a channel and terminates the call. > > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 17 March 2009 15:48 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: > >> Another update - this time (part) good news! Decided to run >> > wancfg_tdmapi again, using the same settings as we always did, and we > can now make external calls. I suspect that whatever BT did yesterday > kicked the circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > > >> However placing an external call into FS isn't as successful, looks >> > like it can't assign a channel and terminates the call. > > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > ------------------------------------------------------------------------ -------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > ------------------------------------------------------------------------ -------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From qulix at mail.ru Thu Mar 19 04:05:23 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Thu, 19 Mar 2009 14:05:23 +0300 Subject: [Freeswitch-users] Good time, people! Message-ID: Hello! Has anybody faced such a problem with xml_curl? 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 1000->********** in context default 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() Oversized file detected [136089828 bytes] 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error encountered! Tcpdump log tells that transaction is allright (xml dialplan is fine and etc) But FS says its oversized =\ what could be wrong? My trunk is : FreeSWITCH Version 1.0.trunk (12573). From codecomplete at free.fr Thu Mar 19 04:54:54 2009 From: codecomplete at free.fr (Gilles) Date: Thu, 19 Mar 2009 12:54:54 +0100 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: References: Message-ID: <7.0.1.0.2.20090319125259.02475630@free.fr> Michael Jerris > There is currently no openzap (sangoma, etc) support on windows, we hope this will be coming soon. I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper too. Would you say the Windows port of Freeswitch is ready to be used commercially, or I should go for a Linux box instead? Thank you. From leon at scarlet-internet.nl Thu Mar 19 04:57:46 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Mar 2009 12:57:46 +0100 Subject: [Freeswitch-users] Proxy media hickups in audio Message-ID: Hi all, I'm still undecided yet whether I need proxy-media or not. As I understand it, the only downside of enabling proxy-media is that early- media is not possible, correct ? (Or are there other reasons why I shouldn't use proxy-media ?) When I disable proxy-media I get little hickups in the audio on the outbound leg of the call, which are not there when proxy-media is enabled. I made pcap dumps on the ethernet interface for both in- and outbound legs, for both settings (proxy-media dis- and enabled), analyzed it in wireshark (no packetloss), but when extracting the audiostreams to .au and listening to it on my laptop, I hear the same hickups. Is there some other setting that may fix the problem of the hickups ? (Is this a known problem?) I'm running fs with -hp argument, am doing no transcoding and am using ALAW codec. Is it helpful if I put the pcap and .au files somewhere ? thanks & regards, Leon From intralanman at freeswitch.org Thu Mar 19 05:25:59 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 19 Mar 2009 08:25:59 -0400 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <22594923.post@talk.nabble.com> References: <22594923.post@talk.nabble.com> Message-ID: <49C239D7.4080101@freeswitch.org> HarryK wrote: > I have Cepstral working. > > Can someone please tell me how to go about having it read RSS feeds? I can > have the dialplan direct it np. But I really dont have a clue how to point > it at an RSS. Any help would be great, ddint find anything in the wiki. > > > have you tried mod_rss? -Ray From freeswitch at gnarg.org Thu Mar 19 05:08:29 2009 From: freeswitch at gnarg.org (freeswitch at gnarg.org) Date: Thu, 19 Mar 2009 13:08:29 +0100 Subject: [Freeswitch-users] FS in Solaris In-Reply-To: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> References: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> Message-ID: <49C235BD.1040905@gnarg.org> Pablo Hernan Saro wrote: > Hi list, > > Any experience building FS in Solaris using Sun Studio? http://www.voiceworks.pl/cypromis/tag/opensolaris/ Chris From stkn at freeswitch.org Thu Mar 19 06:12:53 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Thu, 19 Mar 2009 14:12:53 +0100 Subject: [Freeswitch-users] FS in Solaris In-Reply-To: <49C235BD.1040905@gnarg.org> References: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> <49C235BD.1040905@gnarg.org> Message-ID: <200903191412.53527.stkn@freeswitch.org> Am Thursday 19 March 2009 schrieb freeswitch at gnarg.org: > Pablo Hernan Saro wrote: > > > Hi list, > > > > Any experience building FS in Solaris using Sun Studio? > > http://www.voiceworks.pl/cypromis/tag/opensolaris/ > > Chris > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > There are still some issues we're currently working on. You'll have to add "--disable-visibility" to the configure line to get a working mod_sofia and softtimer. Mod_lua is broken too and i'm still trying to find a fix for that. SFEcurses and SFEncursesw may cause FS to segfault on startup (if those are installed on your system), "--disable-core-libedit-support" will fix that, but you'll loose the line edit and history feature of the FS console. stkn -- Stefan Knoblich Web: http://stkn.techmage.de/ http://oss.axsentis.de/people/stkn/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From brian at freeswitch.org Thu Mar 19 06:14:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 08:14:52 -0500 Subject: [Freeswitch-users] Good time, people! In-Reply-To: References: Message-ID: <1A6B5E65-7091-41EC-9C67-619E17632F71@freeswitch.org> Any reason you're feeding it a 130+ meg file over XML_CURL? /b On Mar 19, 2009, at 6:05 AM, ????... wrote: > Hello! > > Has anybody faced such a problem with xml_curl? > 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing 1000->********** in context default > 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() > Oversized file detected [136089828 bytes] > 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error > encountered! > > Tcpdump log tells that transaction is allright (xml dialplan is fine > and etc) > But FS says its oversized =\ what could be wrong? > > My trunk is : > FreeSWITCH Version 1.0.trunk (12573). From brian at freeswitch.org Thu Mar 19 06:15:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 08:15:51 -0500 Subject: [Freeswitch-users] Proxy media hickups in audio In-Reply-To: References: Message-ID: You shouldn't use it. It has a special use case and I suspect yours isn't it. Are you doing anything with T.38 right now? /b On Mar 19, 2009, at 6:57 AM, Leon de Rooij wrote: > I'm still undecided yet whether I need proxy-media or not. As I > understand it, the only downside of enabling proxy-media is that > early- > media is not possible, correct ? (Or are there other reasons why I > shouldn't use proxy-media ?) From leon at scarlet-internet.nl Thu Mar 19 06:30:10 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Mar 2009 14:30:10 +0100 Subject: [Freeswitch-users] Proxy media hickups in audio In-Reply-To: References: Message-ID: <8D62A4C8-C6DB-4860-B12F-3E02C3C2EE63@scarlet-internet.nl> I'm not using T38 yet, it may be nice in the future, as long faxes over alaw just don't work properly.. And also, there are these hickups now, that I don't have with proxy- media enabled.. On Mar 19, 2009, at 2:15 PM, Brian West wrote: > You shouldn't use it. It has a special use case and I suspect yours > isn't it. Are you doing anything with T.38 right now? > > /b > > On Mar 19, 2009, at 6:57 AM, Leon de Rooij wrote: > >> I'm still undecided yet whether I need proxy-media or not. As I >> understand it, the only downside of enabling proxy-media is that >> early- >> media is not possible, correct ? (Or are there other reasons why I >> shouldn't use proxy-media ?) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pablosaro at gmail.com Thu Mar 19 06:34:58 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Thu, 19 Mar 2009 10:34:58 -0300 Subject: [Freeswitch-users] FS in Solaris In-Reply-To: <200903191412.53527.stkn@freeswitch.org> References: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> <49C235BD.1040905@gnarg.org> <200903191412.53527.stkn@freeswitch.org> Message-ID: <247f8100903190634u63d2e29bm7dd0b74abf1976b3@mail.gmail.com> Well, I guess that is something I can deal with... Actually it is for benchmarking purposes. I was discussing about performance with a colleague, who is a Sr Solaris Engineer, and he recommended me to build FS in Solaris and benchmark it. He ensures that it would be really better due to Fire Engine, the Solaris' networking stack. Thank you guys for your replies. I will look into it, following your guide lines and bothering my colleague :) I hope to get interesting results to share with the community. Regards Pablo On Thu, Mar 19, 2009 at 10:12 AM, Stefan Knoblich wrote: > Am Thursday 19 March 2009 schrieb freeswitch at gnarg.org: >> Pablo Hernan Saro wrote: >> >> > Hi list, >> > >> > Any experience building FS in Solaris using Sun Studio? >> >> http://www.voiceworks.pl/cypromis/tag/opensolaris/ >> >> Chris >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > There are still some issues we're currently working on. > You'll have to add "--disable-visibility" to the configure line to get a working > mod_sofia and softtimer. > Mod_lua is broken too and i'm still trying to find a fix for that. > > SFEcurses and SFEncursesw may cause FS to segfault on startup > (if those are installed on your system), "--disable-core-libedit-support" > will fix that, but you'll loose the line edit and history feature of the FS console. > > > stkn > > -- > Stefan Knoblich > > Web: ? http://stkn.techmage.de/ ? ? ? ?http://oss.axsentis.de/people/stkn/ > Email: stkn at freeswitch.org > IRC: ? ?#freeswitch-de @ irc.freenode.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From michal.bielicki at voiceworks.pl Thu Mar 19 06:41:59 2009 From: michal.bielicki at voiceworks.pl (Michal Bielicki) Date: Thu, 19 Mar 2009 14:41:59 +0100 Subject: [Freeswitch-users] FS in Solaris In-Reply-To: <247f8100903190634u63d2e29bm7dd0b74abf1976b3@mail.gmail.com> References: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> <49C235BD.1040905@gnarg.org> <200903191412.53527.stkn@freeswitch.org> <247f8100903190634u63d2e29bm7dd0b74abf1976b3@mail.gmail.com> Message-ID: <49C24BA7.4060802@voiceworks.pl> If you do a proper side by side test, let me know the results and we will publish them. cheers Michal Pablo Hernan Saro schrieb: > Well, I guess that is something I can deal with... Actually it is for > benchmarking purposes. I was discussing about performance with a > colleague, who is a Sr Solaris Engineer, and he recommended me to > build FS in Solaris and benchmark it. He ensures that it would be > really better due to Fire Engine, the Solaris' networking stack. > Thank you guys for your replies. I will look into it, following your > guide lines and bothering my colleague :) > I hope to get interesting results to share with the community. > Regards > > Pablo > > On Thu, Mar 19, 2009 at 10:12 AM, Stefan Knoblich wrote: > >> Am Thursday 19 March 2009 schrieb freeswitch at gnarg.org: >> >>> Pablo Hernan Saro wrote: >>> >>> >>>> Hi list, >>>> >>>> Any experience building FS in Solaris using Sun Studio? >>>> >>> http://www.voiceworks.pl/cypromis/tag/opensolaris/ >>> >>> Chris >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> There are still some issues we're currently working on. >> You'll have to add "--disable-visibility" to the configure line to get a working >> mod_sofia and softtimer. >> Mod_lua is broken too and i'm still trying to find a fix for that. >> >> SFEcurses and SFEncursesw may cause FS to segfault on startup >> (if those are installed on your system), "--disable-core-libedit-support" >> will fix that, but you'll loose the line edit and history feature of the FS console. >> >> >> stkn >> >> -- >> Stefan Knoblich >> >> Web: http://stkn.techmage.de/ http://oss.axsentis.de/people/stkn/ >> Email: stkn at freeswitch.org >> IRC: #freeswitch-de @ irc.freenode.net >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From michal.bielicki at voiceworks.pl Thu Mar 19 06:43:18 2009 From: michal.bielicki at voiceworks.pl (Michal Bielicki) Date: Thu, 19 Mar 2009 14:43:18 +0100 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: References: <7.0.1.0.2.20090317101702.02701c88@fredshack.com> Message-ID: <49C24BF6.2080207@voiceworks.pl> Michael Jerris schrieb: > There is currently no openzap (sangoma, etc) support on windows, we > hope this will be coming soon. > > Mike > > On Mar 17, 2009, at 5:20 AM, Gilles wrote: > > >> Hello >> >> For single-host settings, getting customers to buy a separate server >> just to run Freeswitch is overkill, so I'm thinking about selling >> just the IVR application to run on Windows. Unless a PCI card is >> available, the FXO connection will be provided by Sangoma's USB >> device. >> >> I'd like some feedback on running Freeswitch on XP and Vista: Is it >> ready for production use? Does it require beefy hardware? >> >> Thank you for any hint. >> > > > You could use Netborder Express with it. From asannucci at gmail.com Thu Mar 19 08:28:36 2009 From: asannucci at gmail.com (Andrea) Date: Thu, 19 Mar 2009 10:28:36 -0500 Subject: [Freeswitch-users] FreeSWITCH Italian Forum Message-ID: Hi, maybe this message can considered off-topic, but i think can be interessing for FreeSWITCH community. There is a new forum on FreeSWITCH for italian people. Please visit www.freeswitch-it.org Any suggest are welcome I hope to do my english a little bit better :) Best Regards - Andrea - From leon at scarlet-internet.nl Thu Mar 19 08:34:14 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Mar 2009 16:34:14 +0100 Subject: [Freeswitch-users] Proxy media hickups in audio In-Reply-To: References: Message-ID: <2E45931E-E495-4DA5-BCA1-8FB545BA2D61@scarlet-internet.nl> Brian, I put two au files here: http://www.ldr.scarlet.nl/ua-to-fs.au http://www.ldr.scarlet.nl/fs-to-mgw.au It's a call from a Siemens SX762 (using ALAW) to FS (no transcoding) which bridges it to a mediagateway. Proxy-media is disabled on the incoming sip_profile. Both au files are extracted from the same pcap file. As you can hear, the second sample sounds bad.. Are there any flags I can set in the incoming or outgoing sip_profile that may fix this ? The server has zero load (it's a Dell R300 with 2.6.24-23-server 64 bit kernel). In fact, this was the only call going through.. FS is version 12163M thanks, Leon On Mar 19, 2009, at 2:15 PM, Brian West wrote: > You shouldn't use it. It has a special use case and I suspect yours > isn't it. Are you doing anything with T.38 right now? > > /b > > On Mar 19, 2009, at 6:57 AM, Leon de Rooij wrote: > >> I'm still undecided yet whether I need proxy-media or not. As I >> understand it, the only downside of enabling proxy-media is that >> early- >> media is not possible, correct ? (Or are there other reasons why I >> shouldn't use proxy-media ?) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Mar 19 08:42:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 10:42:02 -0500 Subject: [Freeswitch-users] Proxy media hickups in audio In-Reply-To: <2E45931E-E495-4DA5-BCA1-8FB545BA2D61@scarlet-internet.nl> References: <2E45931E-E495-4DA5-BCA1-8FB545BA2D61@scarlet-internet.nl> Message-ID: I would have to have the raw pcap to make any sense out of it. /b On Mar 19, 2009, at 10:34 AM, Leon de Rooij wrote: > Brian, > > I put two au files here: > > http://www.ldr.scarlet.nl/ua-to-fs.au > http://www.ldr.scarlet.nl/fs-to-mgw.au > > It's a call from a Siemens SX762 (using ALAW) to FS (no transcoding) > which bridges it to a mediagateway. > Proxy-media is disabled on the incoming sip_profile. > Both au files are extracted from the same pcap file. > > As you can hear, the second sample sounds bad.. Are there any flags I > can set in the incoming or outgoing sip_profile that may fix this ? > > The server has zero load (it's a Dell R300 with 2.6.24-23-server 64 > bit kernel). In fact, this was the only call going through.. > > FS is version 12163M > > thanks, > > Leon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/3c38c007/attachment.html From msc at freeswitch.org Thu Mar 19 08:48:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Mar 2009 08:48:00 -0700 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <8CB7658C3800442-84C-1EAB@WEBMAIL-DY38.sysops.aol.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> <87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com> <8CB7658C3800442-84C-1EAB@WEBMAIL-DY38.sysops.aol.com> Message-ID: <87f2f3b90903190848p3839519ah8fa01361c7dfdf84@mail.gmail.com> > tone_detect! sounds good. > > BTW, was there any errors in those extensions I posted. I modified something > you posted MC. Not at first glance. What did you change? -MC From msc at freeswitch.org Thu Mar 19 09:11:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Mar 2009 09:11:36 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> <49C12CD8.7020203@gmx.net> <11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <87f2f3b90903190911s6082877at1231c27f6a86506@mail.gmail.com> On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron wrote: > So the second issue is possibly known - really could do with a fix or a > workaround for this as we plan to use E1's for all incoming traffic. > > Can anyone shed light on the first problem (extension rings for a > fraction of a second then hangs up) I mentioned below, or is that > possibly part of the same issue? I have experienced this before but I believe it was resolved by having the telco switch protocol dialects which is probably not an option for you. I think your best bet is to use ozmod_libpri and see if the issue is still present. -MC From msc at freeswitch.org Thu Mar 19 09:14:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Mar 2009 09:14:36 -0700 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: <7.0.1.0.2.20090319125259.02475630@free.fr> References: <7.0.1.0.2.20090319125259.02475630@free.fr> Message-ID: <87f2f3b90903190914o2098b2c2l3b05812b30d38880@mail.gmail.com> On Thu, Mar 19, 2009 at 4:54 AM, Gilles wrote: > Michael Jerris > There is currently no openzap (sangoma, etc) support > on windows, we ?hope this will be coming soon. > > I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper too. > > Would you say the Windows port of Freeswitch is ready to be used > commercially, or I should go for a Linux box instead? Is there a compelling reason to use a Windows machine? If it's a matter of your comfort level with the OS then that's a pretty good reason, especially if you're the one doing support. :) However, I think all of our "power users" are running Linux, and most of them are using 64 bit CentOS on 64 bit hardware. I suppose it all boils down to what you hope to accomplish. The larger the install, the more I would recommend Linux over Windows. -MC From mike at jerris.com Thu Mar 19 09:34:55 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 19 Mar 2009 12:34:55 -0400 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: <7.0.1.0.2.20090319125259.02475630@free.fr> References: <7.0.1.0.2.20090319125259.02475630@free.fr> Message-ID: <40265B71-2148-4296-956B-AA475CC8C267@jerris.com> On Mar 19, 2009, at 7:54 AM, Gilles wrote: > Michael Jerris > There is currently no openzap (sangoma, etc) support > on windows, we hope this will be coming soon. > > I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper > too. > > Would you say the Windows port of Freeswitch is ready to be used > commercially, or I should go for a Linux box instead? > Generally I won't make a recommendation on things like this beyond saying that regardless of what you choose, you should properly test it and verify that it is stable and suitable for your purposes. Windows may work fine for you, it may not, the same goes for linux or anything else, the only way to know for sure is to try it and see if it fits your needs or not. Mike From Mark.Tabron at rnid-typetalk.org.uk Thu Mar 19 09:54:31 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Thu, 19 Mar 2009 16:54:31 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL><49C12CD8.7020203@gmx.net><11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903190911s6082877at1231c27f6a86506@mail.gmail.com> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765381@tt-mail.RNID.TYPETALK.LOCAL> Thanks, found an install guide on the FS Wiki for libpri - will get the server cloned then install and test. Shall report back. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 19 March 2009 16:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron wrote: > So the second issue is possibly known - really could do with a fix or a > workaround for this as we plan to use E1's for all incoming traffic. > > Can anyone shed light on the first problem (extension rings for a > fraction of a second then hangs up) I mentioned below, or is that > possibly part of the same issue? I have experienced this before but I believe it was resolved by having the telco switch protocol dialects which is probably not an option for you. I think your best bet is to use ozmod_libpri and see if the issue is still present. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- From mszlazak at aol.com Thu Mar 19 10:14:38 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 19 Mar 2009 13:14:38 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <87f2f3b90903190848p3839519ah8fa01361c7dfdf84@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com><35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com><8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com><87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com><87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com><8CB7658C3800442-84C-1EAB@WEBMAIL-DY38.sysops.aol.com> <87f2f3b90903190848p3839519ah8fa01361c7dfdf84@mail.gmail.com> Message-ID: <8CB76CD515B680C-418-AB@WEBMAIL-DZ07.sysops.aol.com> I put this after the "vmd" tag to check vmd with tones found on this page http://en.wikipedia.org/wiki/Special_information_tone I converted them over with Audacity to wav files and "vmd" worked in finding a "beep" but the format was wrong for FS. However, after I switch the format of the audio files to something FS likes then vmd would not detect the tones. Is there some good test tones for the U.S phone system I could use to check both mod_vmd and tone_detect? Thanks. Mark. -----Original Message-----From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Thu, 19 Mar 2009 8:48 am Subject: Re: [Freeswitch-users] Is mod vmd working? > tone_detect! sounds good. > > BTW, was there any errors in those extensions I posted. I modified something > you posted MC. Not at first glance. What did you change? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/0bf36af8/attachment-0001.html From lukasz at czerpak.eu Thu Mar 19 11:22:07 2009 From: lukasz at czerpak.eu (=?UTF-8?B?xYF1a2FzeiBDemVycGFr?=) Date: Thu, 19 Mar 2009 19:22:07 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) Message-ID: <49C28D4F.4040307@czerpak.eu> Hi, I have some troubles with provider configuration. The are warnings in logs: 2009-03-19 19:02:48 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 40 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. My provider uses SER and Cisco hardware (Cisco-SIPGateway/IOS-12.x). Moreover, above warning appears only when someone calls to FreeSWITCH (voice quality is poor) - connection from FreeSWITCH works without any warnings and has perfect quality. Other my providers works perfect. Is there any solution of this problem? regards, Lukasz From brian at freeswitch.org Thu Mar 19 11:28:44 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 13:28:44 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C28D4F.4040307@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> Message-ID: <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> Try: /b On Mar 19, 2009, at 1:22 PM, ?ukasz Czerpak wrote: > Is there any solution of this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/88211d93/attachment.html From lukasz at czerpak.eu Thu Mar 19 11:52:22 2009 From: lukasz at czerpak.eu (=?UTF-8?B?xYF1a2FzeiBDemVycGFr?=) Date: Thu, 19 Mar 2009 19:52:22 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> References: <49C28D4F.4040307@czerpak.eu> <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> Message-ID: <49C29466.5040708@czerpak.eu> Brian West wrote: > Try: > > > * Unfortunately there is no difference when it is set to 'scrooge' or other value :( regards, Lukasz From brian at freeswitch.org Thu Mar 19 11:58:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 13:58:06 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C29466.5040708@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> <49C29466.5040708@czerpak.eu> Message-ID: <3B28A868-3ACE-4BFB-80D0-6D89E7384BDF@freeswitch.org> what rev are you on? /b On Mar 19, 2009, at 1:52 PM, ?ukasz Czerpak wrote: >> * > > Unfortunately there is no difference when it is set to 'scrooge' or > other value :( > > > regards, > Lukasz From kristian.kielhofner at gmail.com Thu Mar 19 11:59:03 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 19 Mar 2009 14:59:03 -0400 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C28D4F.4040307@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> Message-ID: <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> Hi, This is a known issue with some of these platforms but for completeness can you send the actual SDP? 2009/3/19 ?ukasz Czerpak : > Hi, > > I have some troubles with provider configuration. The are warnings in logs: > > 2009-03-19 19:02:48 [WARNING] mod_sofia.c:739 sofia_read_frame() We were > told to use ptime 20 but what they meant to say was 40 > > This issue has so far been identified to happen on the following broken > platforms/devices: > > Linksys/Sipura aka Cisco > > > ShoreTel > > > Sonus/L3 > > > We will try to fix it but some of the devices on this list are so broken > who knows what will happen.. > > My provider uses SER and Cisco hardware (Cisco-SIPGateway/IOS-12.x). > > Moreover, above warning appears only when someone calls to FreeSWITCH > (voice quality is poor) - connection from FreeSWITCH works without any > warnings and has perfect quality. > > Other my providers works perfect. > > Is there any solution of this problem? > > regards, > Lukasz > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From lukasz at czerpak.eu Thu Mar 19 12:20:18 2009 From: lukasz at czerpak.eu (=?ISO-8859-2?Q?=A3ukasz_Czerpak?=) Date: Thu, 19 Mar 2009 20:20:18 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> Message-ID: <49C29AF2.2000703@czerpak.eu> Kristian Kielhofner wrote: > Hi, > > This is a known issue with some of these platforms but for > completeness can you send the actual SDP? My voip configuration: +------------+ +----------+ | ROUTER | LAN | Linksys | NET----| with |---------| PAP2T-EU | | FreeSWITCH | | | +------------+ +----------+ Linksys has changed rtp packet size to 0.020 On FreeSWITCH i am using g729 (http://freehg.org/u/deepwalker/fs_g729/) - for testing only. SDP logs are below, full session's log is in attachment: 2009-03-19 20:07:02 [DEBUG] sofia.c:2806 sofia_handle_sip_i_state() Channel sofia/external/0607xxxxxx at 217.11.128.50 entering state [received] 2009-03-19 20:07:02 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=CiscoSystemsSIP-GW-UserAgent 9221 3981 IN IP4 217.11.128.50 s=SIP Call c=IN IP4 217.11.128.50 t=0 0 m=audio 18292 RTP/AVP 18 4 2 98 99 0 8 3 100 101 19 c=IN IP4 217.11.128.50 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=rtpmap:2 G726-32/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:99 G726-16/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=direction:active ... 2009-03-19 20:07:02 [INFO] mod_sofia.c:1351 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1237472006 1237472007 IN IP4 89.79.191.29 s=FreeSWITCH c=IN IP4 89.79.191.29 t=0 0 m=audio 17616 RTP/AVP 18 101 19 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20 a=sendrecv ... 2009-03-19 20:07:03 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 121150237 121150237 IN IP4 192.168.1.250 s=- c=IN IP4 192.168.1.250 t=0 0 m=audio 16438 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 ... 2009-03-19 20:07:03 [DEBUG] mod_sofia.c:503 sofia_answer_channel() Local SDP sofia/external/0607320038 at 217.11.128.50: v=0 o=FreeSWITCH 1237472006 1237472008 IN IP4 89.79.191.29 s=FreeSWITCH c=IN IP4 89.79.191.29 t=0 0 m=audio 17616 RTP/AVP 18 101 19 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20 a=sendrecv ... 2009-03-19 20:07:04 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=CiscoSystemsSIP-GW-UserAgent 9221 3981 IN IP4 217.11.128.50 s=SIP Call c=IN IP4 217.11.128.50 t=0 0 m=audio 18292 RTP/AVP 18 19 101 100 c=IN IP4 217.11.128.50 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=ptime:60 regards, Lukasz -------------- next part -------------- A non-text attachment was scrubbed... Name: session2.log Type: text/x-log Size: 40912 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/816d2158/attachment-0001.bin From brian at freeswitch.org Thu Mar 19 12:27:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 14:27:56 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C29AF2.2000703@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> Message-ID: Well you can't have ptime 60 one way and 20 the other it just won't work. Also I can't even think that this illegal codec was even tested at 60ms... Try it with ulaw and see what it does. or only allow G729 at 60i and see what it does. /b On Mar 19, 2009, at 2:20 PM, ?ukasz Czerpak wrote: > Linksys has changed rtp packet size to 0.020 > On FreeSWITCH i am using g729 (http://freehg.org/u/deepwalker/ > fs_g729/) - for testing only. > > SDP logs are below, full session's log is in attachment: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/79d48f4d/attachment.html From steve.d.ward at gmail.com Thu Mar 19 12:42:55 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Thu, 19 Mar 2009 15:42:55 -0400 Subject: [Freeswitch-users] not hanging up Message-ID: <4ea6e8f20903191242vb56bcaetb7acd4853b215b0e@mail.gmail.com> I have phones registered to a FS box, and an * box. There is a sip trunk between the two boxes. A phone on my * (54321) calls a FS phone (12345); if I hang up the * phone while it's still ringing, this is what I get on the sip trace on FS: ... 2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/12345 at 11.2.22.45! recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950: ------------------------------------------------------------------------ CANCEL sip:12345 at b-pbx-sip-3.abc.xyz.netSIP/2.0 Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport From: "Steve" >;tag=as25193d44 To: > Call-ID: 0c0614d866a62841546cbf3340224682 at 11.2.22.45 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 328 bytes to udp/[11.2.22.45]:5060 at 19:05:44.313572: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport=5060 From: "Steve" >;tag=as25193d44 To: >;tag=c5Z8Q1e93p7KD Call-ID: 0c0614d866a62841546cbf3340224682 at 11.2.22.45 CSeq: 103 CANCEL Content-Length: 0 -------------------------------------------------------- The effect is that the FS keeps on ringing - it doesn't detect the hangup. When I call from a FS phone (1000) to another FS phone (12345), and I hang up the calling phone while it's still ringing, this is what I get on the sip trace: ... send 425 bytes to udp/[11.2.56.106]:63054 at 19:15:29.737163: ------------------------------------------------------------------------ CANCEL sip:12345 at 11.2.56.106:63054;rinstance=64e968d7a1317bc3 SIP/2.0 Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a Max-Forwards: 69 From: "Extension 1000" >;tag=meK8yUgpgU2Zc To: Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 ------------------------------------------------------------------------ recv 427 bytes from udp/[11.2.56.106]:63054 at 19:15:29.838863: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a Contact: To: ;tag=db12c87a From: "Extension 1000" >;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ------------------------------------------------------------------------ recv 376 bytes from udp/[11.2.56.106]:63054 at 19:15:29.839334: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a To: ;tag=db12c87a From: "Extension 1000" >;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ... It works just fine. Any ideas? I'm not sure where to go with this. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/67594f9f/attachment.html From lukasz at czerpak.eu Thu Mar 19 12:43:51 2009 From: lukasz at czerpak.eu (=?UTF-8?B?xYF1a2FzeiBDemVycGFr?=) Date: Thu, 19 Mar 2009 20:43:51 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <3B28A868-3ACE-4BFB-80D0-6D89E7384BDF@freeswitch.org> References: <49C28D4F.4040307@czerpak.eu> <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> <49C29466.5040708@czerpak.eu> <3B28A868-3ACE-4BFB-80D0-6D89E7384BDF@freeswitch.org> Message-ID: <49C2A077.8080409@czerpak.eu> Brian West wrote: > what rev are you on? > trunk - ~2009-03-15 21:00 regards, ?ukasz From lukasz at czerpak.eu Thu Mar 19 13:06:31 2009 From: lukasz at czerpak.eu (=?ISO-8859-2?Q?=A3ukasz_Czerpak?=) Date: Thu, 19 Mar 2009 21:06:31 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> Message-ID: <49C2A5C7.10205@czerpak.eu> Brian West wrote: > Well you can't have ptime 60 one way and 20 the other it just won't > work. Also I can't even think that this illegal codec was even tested > at 60ms... Try it with ulaw and see what it does. or only allow > G729 at 60i and see what it does. > I've just tested G729 at 60i and everything works perfect - thank you very much. I didn't test ulaw. What is wrong - my provider is incompatible with specification or FreeSWITCH has problem with codec negotiation? Is there any possibility to force codec for specific gateway/provider? regards Lukasz From brian at freeswitch.org Thu Mar 19 13:31:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 15:31:18 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C2A5C7.10205@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> <49C2A5C7.10205@czerpak.eu> Message-ID: The issue I seen was they invite to you with NO ptime which indicates 20ms, they should invite with ptime:60 if they want 60. /b On Mar 19, 2009, at 3:06 PM, ?ukasz Czerpak wrote: > > I've just tested G729 at 60i and everything works perfect - thank you > very > much. I didn't test ulaw. > What is wrong - my provider is incompatible with specification or > FreeSWITCH has problem with codec negotiation? > > Is there any possibility to force codec for specific gateway/provider? > > regards > Lukasz From qulix at mail.ru Thu Mar 19 13:48:32 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Thu, 19 Mar 2009 23:48:32 +0300 Subject: [Freeswitch-users] =?koi8-r?b?R29vZCB0aW1lLCBwZW9wbGUh?= Message-ID: Thats the thing!! Im using tcpdump to watch for packets - and i dont see any mistakes =\ The xml i sent is allright, its like a piece from my static worked xml dialplan. But I cant understand why does FS recognise it as a 130+ mb file :D Maybe i need to update s0mthing?) Brian West ?????: > Any reason you're feeding it a 130+ meg file over XML_CURL? > > /b > > On Mar 19, 2009, at 6:05 AM, ????... wrote: > >> Hello! >> >> Has anybody faced such a problem with xml_curl? >> 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 1000->********** in context default >> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() >> Oversized file detected [136089828 bytes] >> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error >> encountered! >> >> Tcpdump log tells that transaction is allright (xml dialplan is fine >> and etc) >> But FS says its oversized =\ what could be wrong? >> >> My trunk is : >> FreeSWITCH Version 1.0.trunk (12573). > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Mar 19 13:52:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 15:52:22 -0500 Subject: [Freeswitch-users] Good time, people! In-Reply-To: References: Message-ID: <9A3D7EE5-FA78-4C64-9868-6E46AB12ED5A@freeswitch.org> can you do a raw request with wget? /b On Mar 19, 2009, at 3:48 PM, ????... wrote: > Thats the thing!! > Im using tcpdump to watch for packets - and i dont see any mistakes =\ > The xml i sent is allright, its like a piece from my static worked > xml dialplan. > > But I cant understand why does FS recognise it as a 130+ mb file :D > Maybe i need to update s0mthing?) From lukasz at czerpak.eu Thu Mar 19 14:03:13 2009 From: lukasz at czerpak.eu (=?UTF-8?B?xYF1a2FzeiBDemVycGFr?=) Date: Thu, 19 Mar 2009 22:03:13 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> <49C2A5C7.10205@czerpak.eu> Message-ID: <49C2B311.2040105@czerpak.eu> Brian West wrote: > The issue I seen was they invite to you with NO ptime which indicates > 20ms, they should invite with ptime:60 if they want 60. > I see but there is any solution to bypass this provider's "incompatibility"? I want to stay with this provider anyway - he has very good quality and nice prices ;) thanks and regards, ?ukasz From brian at freeswitch.org Thu Mar 19 14:08:46 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 16:08:46 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C2B311.2040105@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> <49C2A5C7.10205@czerpak.eu> <49C2B311.2040105@czerpak.eu> Message-ID: Well you said you were using G.729 for testing... when you're clearly not... but I told you already how to fix it... for that IP or peer G729 at 60i /b On Mar 19, 2009, at 4:03 PM, ?ukasz Czerpak wrote: > I see but there is any solution to bypass this provider's > "incompatibility"? I want to stay with this provider anyway - he has > very good quality and nice prices ;) > > thanks and regards, > ?ukasz From qulix at mail.ru Thu Mar 19 14:28:45 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Fri, 20 Mar 2009 00:28:45 +0300 Subject: [Freeswitch-users] =?koi8-r?b?R29vZCB0aW1lLCBwZW9wbGUh?= Message-ID: Its not the easy thing. But what I can do is to attach here full tcpdump log, with all packets. Brian West ?????: > can you do a raw request with wget? > > /b > > On Mar 19, 2009, at 3:48 PM, ????... wrote: > >> Thats the thing!! >> Im using tcpdump to watch for packets - and i dont see any mistakes =\ >> The xml i sent is allright, its like a piece from my static worked >> xml dialplan. >> >> But I cant understand why does FS recognise it as a 130+ mb file :D >> Maybe i need to update s0mthing?) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sprice at gmail.com Thu Mar 19 14:33:58 2009 From: sprice at gmail.com (SP) Date: Thu, 19 Mar 2009 16:33:58 -0500 Subject: [Freeswitch-users] Good time, people! In-Reply-To: References: Message-ID: <7e2ac3270903191433h35e043d3l8e948690599a609@mail.gmail.com> Are you setting a Content Length header in the HTTP response?? 2009/3/19 ????... : > Thats the thing!! > Im using tcpdump to watch for packets - and i dont see any mistakes =\ > The xml i sent is allright, its like a piece from my static worked xml dialplan. > > But I cant understand why does FS recognise it as a 130+ mb file :D > Maybe i need to update s0mthing?) > > Brian West ?????: >> Any reason you're feeding it a 130+ meg file over XML_CURL? >> >> /b >> >> On Mar 19, 2009, at 6:05 AM, ????... wrote: >> >>> Hello! >>> >>> Has anybody faced such a problem with xml_curl? >>> 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>> Processing 1000->********** in context default >>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() >>> Oversized file detected [136089828 bytes] >>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error >>> encountered! >>> >>> Tcpdump log tells that transaction is allright (xml dialplan is fine >>> and etc) >>> But FS says its oversized =\ what could be wrong? >>> >>> My trunk is : >>> FreeSWITCH Version 1.0.trunk (12573). >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From qulix at mail.ru Thu Mar 19 14:45:17 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Fri, 20 Mar 2009 00:45:17 +0300 Subject: [Freeswitch-users] =?koi8-r?b?R29vZCB0aW1lLCBwZW9wbGUh?= Message-ID: As I see theres only : Content-Type: text/html; charset=utf-8 But no Content Length SP ?????: > Are you setting a Content Length header in the HTTP response?? > > 2009/3/19 ????... : >> Thats the thing!! >> Im using tcpdump to watch for packets - and i dont see any mistakes =\ >> The xml i sent is allright, its like a piece from my static worked xml dialplan. >> >> But I cant understand why does FS recognise it as a 130+ mb file :D >> Maybe i need to update s0mthing?) >> >> Brian West ?????: >>> Any reason you're feeding it a 130+ meg file over XML_CURL? >>> >>> /b >>> >>> On Mar 19, 2009, at 6:05 AM, ????... wrote: >>> >>>> Hello! >>>> >>>> Has anybody faced such a problem with xml_curl? >>>> 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>>> Processing 1000->********** in context default >>>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() >>>> Oversized file detected [136089828 bytes] >>>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error >>>> encountered! >>>> >>>> Tcpdump log tells that transaction is allright (xml dialplan is fine >>>> and etc) >>>> But FS says its oversized =\ what could be wrong? >>>> >>>> My trunk is : >>>> FreeSWITCH Version 1.0.trunk (12573). >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From codecomplete at free.fr Thu Mar 19 15:45:07 2009 From: codecomplete at free.fr (Gilles) Date: Thu, 19 Mar 2009 23:45:07 +0100 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? Message-ID: <7.0.1.0.2.20090319232207.0242a5d8@fredshack.com> (sorry for the broken thread: I don't know how to avoid this when answering through the digest version of the mailing list) Michael Jerris > You could use Netborder Express with it. Thanks for the tip. I didn't know this device. I'm not sure I understand the difference between this PCI card and other Sangoma PCI cards that offer an FXO port, though :-/ mercutioviz > Is there a compelling reason to use a Windows machine? Yes. I'd like to offer a really cheap solution for those customers who don't mind using their workstation as Freeswitch IVR server, so I can just provide a Linksys VoIP gateway and the software for Windows, and they're ready to go. I'll go ahead and play with the Windows port of Freeswitch, and see how it goes. Thank you. From mszlazak at aol.com Thu Mar 19 16:35:47 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 19 Mar 2009 19:35:47 -0400 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: <7.0.1.0.2.20090319232207.0242a5d8@fredshack.com> References: <7.0.1.0.2.20090319232207.0242a5d8@fredshack.com> Message-ID: <8CB7702907EC8A8-1188-10E9@WEBMAIL-DY32.sysops.aol.com> I'm doing what you want to do and using SPA3102.? It's much easier to get someone to try it this way when dealing with small mom and pop size business.? Haven't tried higher concurrent call volumes with some of the PCI cards mentioned. If you haven't done this already, my advice is first to see whether there is enough of a market for you and what the issues will be with potential customers before investing to much time on the technical side. That means putting on your suit on and visiting businesses for a few months with a notebook. I'll tell you some things, first their maybe many businesses that don't want an internet connection and don't even bother mentioning voip if you want to resell that service. The rest you'll find out. Good luck. Mark. -----Original Message----- From: Gilles To: freeswitch-users at lists.freeswitch.org Sent: Thu, 19 Mar 2009 3:45 pm Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? (sorry for the broken thread: I don't know how to avoid this when answering through the digest version of the mailing list) Michael Jerris > You could use Netborder Express with it. Thanks for the tip. I didn't know this device. I'm not sure I understand the difference between this PCI card and other Sangoma PCI cards that offer an FXO port, though :-/ mercutioviz > Is there a compelling reason to use a Windows machine? Yes. I'd like to offer a really cheap solution for those customers who don' t mind using their workstation as Freeswitch IVR server, so I can just provide a Linksys VoIP gateway and the software for Windows, and they're ready to go. I'll go ahead and play with the Windows port of Freeswitch, and see how it goes. Thank you. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/13415cb2/attachment.html From switchserver at gmail.com Thu Mar 19 19:53:29 2009 From: switchserver at gmail.com (HarryK) Date: Thu, 19 Mar 2009 19:53:29 -0700 (PDT) Subject: [Freeswitch-users] Skypiax and no-audio (NAT) issue Message-ID: <22613477.post@talk.nabble.com> Ok I got Skypiax working just fine but there is no audio either way when I say call into a conf using the Skype username. I had this no audio problem with NAT when I first setup FreeSWITCH and solved it by using "Scenario 2" from this wiki page... http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios Now I'm stumped because I THINK I have to traverse NAT once again but not seeing anyway to tell Skypiax that it must use the "doublenat" profile! If thats even close to the solution I'm not sure. The call is ending up in the proper conf, just no audio. I've confirmed this. Does someone have any experience with what I'm dealing with? Thanks -- View this message in context: http://www.nabble.com/Skypiax-and-no-audio-%28NAT%29-issue-tp22613477p22613477.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From switchserver at gmail.com Thu Mar 19 19:55:58 2009 From: switchserver at gmail.com (HarryK) Date: Thu, 19 Mar 2009 19:55:58 -0700 (PDT) Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <49C239D7.4080101@freeswitch.org> References: <22594923.post@talk.nabble.com> <49C239D7.4080101@freeswitch.org> Message-ID: <22613496.post@talk.nabble.com> I guess that's why they call us noobs! heh ;) Working perfectly, thank you!! Raymond Chandler-2 wrote: > > HarryK wrote: >> I have Cepstral working. >> >> Can someone please tell me how to go about having it read RSS feeds? I >> can >> have the dialplan direct it np. But I really dont have a clue how to >> point >> it at an RSS. Any help would be great, ddint find anything in the wiki. >> >> >> > have you tried mod_rss? > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22613496.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Thu Mar 19 20:26:53 2009 From: dujinfang at gmail.com (seven) Date: Fri, 20 Mar 2009 11:26:53 +0800 Subject: [Freeswitch-users] Skypiax and no-audio (NAT) issue In-Reply-To: <22613477.post@talk.nabble.com> References: <22613477.post@talk.nabble.com> Message-ID: <86005AC0-14AE-4505-B97D-2A6F3F2B0A22@gmail.com> I don't think no sound is caused by NAT, better to check sound driver and configuration. On Mar 20, 2009, at 10:53 AM, HarryK wrote: > > Ok I got Skypiax working just fine but there is no audio either way > when I > say call into a conf using the Skype username. > > I had this no audio problem with NAT when I first setup FreeSWITCH and > solved it by using "Scenario 2" from this wiki page... > > http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios > > Now I'm stumped because I THINK I have to traverse NAT once again > but not > seeing anyway to tell Skypiax that it must use the "doublenat" > profile! If > thats even close to the solution I'm not sure. The call is ending up > in the > proper conf, just no audio. I've confirmed this. > > Does someone have any experience with what I'm dealing with? > > > Thanks > -- > View this message in context: http://www.nabble.com/Skypiax-and-no-audio-%28NAT%29-issue-tp22613477p22613477.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mattdfong at gmail.com Thu Mar 19 20:46:45 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 20 Mar 2009 10:46:45 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> Message-ID: <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> Hi Anthony, I installed the patch, but I don't think it accomplishes what I want. I want the opposite, I want the fifo caller to continue along with the dialplan after the agent (consumer) is finished with servicing the call. This might be useful in situations where there could be an IVR recording customer satisfaction results of the agent servicing the call. As is, FS ends the caller's channel after finishing up in the fifo (ie, agent (consumer) disconnects or hangsup)--there should be life after s/he has been serviced by an agent (preferably continuing on in the dialplan). If I'm confused and missing something obvious, please correct me... Thanks --matt 2009/3/19 Anthony Minessale > This is the patch > > http://jira.freeswitch.org/browse/MODAPP-237 > > it's not added yet. > > > 2009/3/18 Matthew Fong > > I upgraded to >> FreeSWITCH Version 1.0.trunk (12654M) >> >> but caller is still being hungup (and not continuing on with dialplan) >> after agent disconnect with hangup_after_bridge=false >> >> Is there a separate patch I need to apply? Thanks. >> >> --matt >> >> >> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >> >>> Hi Anthony, thanks for the reply. >>> I've searched thru jira, and didn't find anything when searching for fifo >>> that was recently updated or related, except >>> >>> http://jira.freeswitch.org/browse/MODAPP-189 >>> >>> and I'm not sure if this does what I need. Was this what you were >>> referring to? Thanks. >>> >>> --matt >>> >>> 2009/3/17 Anthony Minessale >>> >>> there is a patch in jira that will implement this feature about to be >>>> added >>>> >>>> >>>> 2009/3/17 Matthew Fong >>>> >>>>> I apologize if this is a double post to -dev. I'm not sure why I don't >>>>> see my message appearing, so I'm going to try again in the -user list (first >>>>> timer posting here ;). >>>>> >>>>> I have a situation where it would be useful for a caller not to be >>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>> situation. It would be preferable if the caller channel went further along >>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>> minimal. I also tried executing fifo in a lua app and >>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>> could be done as a feature enhancement? Thanks. >>>>> >>>>> --matt >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/5e72790d/attachment.html From carlos.talbot at gmail.com Thu Mar 19 22:02:04 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Fri, 20 Mar 2009 00:02:04 -0500 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <22594923.post@talk.nabble.com> References: <22594923.post@talk.nabble.com> Message-ID: <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> I wrote this wiki page a while back. Did it help? http://wiki.freeswitch.org/wiki/Mod_rss On Thu, Mar 19, 2009 at 2:41 AM, HarryK wrote: > > I have Cepstral working. > > Can someone please tell me how to go about having it read RSS feeds? I can > have the dialplan direct it np. But I really dont have a clue how to point > it at an RSS. Any help would be great, ddint find anything in the wiki. > > > -- > View this message in context: > http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22594923.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/2fd8228d/attachment.html From brian at freeswitch.org Thu Mar 19 22:08:23 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Mar 2009 00:08:23 -0500 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> Message-ID: http://jira.freeswitch.org/browse/MODASRTTS-11 Might wanna know about that issue also :) /b On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote: > I wrote this wiki page a while back. Did it help? > > http://wiki.freeswitch.org/wiki/Mod_rss -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/935dabec/attachment.html From lukasz at czerpak.eu Fri Mar 20 01:05:40 2009 From: lukasz at czerpak.eu (=?ISO-8859-2?Q?=A3ukasz_Czerpak?=) Date: Fri, 20 Mar 2009 09:05:40 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> Message-ID: <49C34E54.9000000@czerpak.eu> Brian West pisze: > Well you can't have ptime 60 one way and 20 the other it just won't > work. Also I can't even think that this illegal codec was even tested > at 60ms... Try it with ulaw and see what it does. or only allow > G729 at 60i and see what it does. > I made some tests with ulaw with success. There is no problem with ptime negotiation. Will the g729 codec be fully (not passthrough) supported in FreeSWITCH? regards, -- ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] From andy at fabulous4.co.uk Fri Mar 20 01:40:41 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Fri, 20 Mar 2009 08:40:41 -0000 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> Message-ID: <153785972B6D42C99E8CAF7DE2F1A145@wsandy> Thanks Brian, I've upgraded to 1.0.3 and things seem a little better but I'm still loosing the gateway connection intermittently. I rebuilt the config on upgrade, is there any possibility I've missed something? Is there a keep-alive setting for a gateway or a re-connect after x or something. Many thanks for your help. Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 18 March 2009 14:08 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Losing Gateway registration Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/bd4029fd/attachment-0001.html From ludovic.fouquet at bewan.com Fri Mar 20 03:11:08 2009 From: ludovic.fouquet at bewan.com (ludovic) Date: Fri, 20 Mar 2009 11:11:08 +0100 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: <246296A5-851A-4859-BCA9-05E2415A20EA@freeswitch.org> References: <49BE928D.3090509@bewan.com> <49BF87E5.5090809@bewan.com> <49BFE901.3070709@bewan.com> <246296A5-851A-4859-BCA9-05E2415A20EA@freeswitch.org> Message-ID: <49C36BBC.1060708@bewan.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/f8f5ac6a/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: bewan100.jpg Type: image/jpeg Size: 3963 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/f8f5ac6a/attachment.jpg From anthony.minessale at gmail.com Fri Mar 20 05:04:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Mar 2009 07:04:09 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> Message-ID: <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> The agent could transfer the caller to another extension. 2009/3/19 Matthew Fong > Hi Anthony, > I installed the patch, but I don't think it accomplishes what I want. > > I want the opposite, I want the fifo caller to continue along with the > dialplan after the agent (consumer) is finished with servicing the call. > This might be useful in situations where there could be an IVR recording > customer satisfaction results of the agent servicing the call. As is, FS > ends the caller's channel after finishing up in the fifo (ie, agent > (consumer) disconnects or hangsup)--there should be life after s/he has been > serviced by an agent (preferably continuing on in the dialplan). > > If I'm confused and missing something obvious, please correct me... Thanks > > --matt > > > > 2009/3/19 Anthony Minessale > > This is the patch >> >> http://jira.freeswitch.org/browse/MODAPP-237 >> >> it's not added yet. >> >> >> 2009/3/18 Matthew Fong >> >> I upgraded to >>> FreeSWITCH Version 1.0.trunk (12654M) >>> >>> but caller is still being hungup (and not continuing on with dialplan) >>> after agent disconnect with hangup_after_bridge=false >>> >>> Is there a separate patch I need to apply? Thanks. >>> >>> --matt >>> >>> >>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>> >>>> Hi Anthony, thanks for the reply. >>>> I've searched thru jira, and didn't find anything when searching for >>>> fifo that was recently updated or related, except >>>> >>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>> >>>> and I'm not sure if this does what I need. Was this what you were >>>> referring to? Thanks. >>>> >>>> --matt >>>> >>>> 2009/3/17 Anthony Minessale >>>> >>>> there is a patch in jira that will implement this feature about to be >>>>> added >>>>> >>>>> >>>>> 2009/3/17 Matthew Fong >>>>> >>>>>> I apologize if this is a double post to -dev. I'm not sure why I don't >>>>>> see my message appearing, so I'm going to try again in the -user list (first >>>>>> timer posting here ;). >>>>>> >>>>>> I have a situation where it would be useful for a caller not to be >>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>> situation. It would be preferable if the caller channel went further along >>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>> minimal. I also tried executing fifo in a lua app and >>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>> could be done as a feature enhancement? Thanks. >>>>>> >>>>>> --matt >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/51f08e52/attachment-0001.html From mark at markehle.net Fri Mar 20 05:29:18 2009 From: mark at markehle.net (Mark) Date: Fri, 20 Mar 2009 08:29:18 -0400 Subject: [Freeswitch-users] (OT) SPA-922 unlock Message-ID: <20090320082918.7060959me3mtzbxc@gateway.ehle.homelinux.org> I was never able to unlock these phones using some sort of sniffing/back door stuff. I was, however, lucky enough to have the previous owner discover the password. If it helps anybody, the now-defunct Webnet Global Communications (where the phones came from at a bankruptcy auction) used the username Admin and password 'webnet'. Thanks to all who provided information on this endeavor. Library Mark Quoting Gabriel Kuri : > On the slight chance they're not doing remote provisioning and the phone > is just simply locked with a username/password, you'll need to feed the > phone a TFTP server via DHCP Option 66 and setup a config file on that > tftp server with the name spa922.cfg. > > Contact me off list about generating a config file for the phone. > > Gabe > > Mark wrote: >> I did unplug the ethernet cable. I have never been able to make the >> IVR work on any of the Linksys phones that I have. I must be doing >> something wrong. >> >> I will try to sniff the traffic on the phone when I start it up. I >> will report back when I do. >> >> Thanks so much - >> >> Library Mark >> >> Quoting "Gabriel Kuri" : >> >>> I believe you need to make sure the Ethernet cable is unplugged from the >>> phone when trying to dial that string. >>> >>> Now I've never tried this, but it should theoretically be possible ... >>> >>> Sniff the traffic of the phone and see where it's attempting to pickup >>> the config file. Then setup a local network with your own DNS server, >>> and re-direct the phone (via DNS) to your own web server (assuming it's >>> picking up the config via http) and have a config file on the web server >>> with a username and password you specify to reset the config and get >>> into the phone. Let's hope they didn't setup the phone to provision via >>> https, otherwise you're really SOL >>> >>> If you need help generating a config for the phone, with Linksys' >>> special config tool, contact me offlist. >>> >>> Gabe >>> >>> Mark wrote: >>>> Sadly, ****73738# does not work. >>>> >>>> Is there a jumper on the board or some other hardware fix for this? >>>> >>>> Quoting "Gabriel Kuri" : >>>> >>>>> Have you tried resetting the phone via the built-in IVR menu? >>>>> >>>>> Pick up the handset and dial ****73738# >>>>> >>>>> This should reset the phone to factory defaults, assuming that company >>>>> didn't lock this feature out. >>>>> >>>>> Gabe >>>>> >>>>> >>>>> >>>>> Mark wrote: >>>>>> Hello, folks - I hope that I can reach someone who knows the answer to >>>>>> this one: >>>>>> >>>>>> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >>>>>> locked by Webnet global Communications. From what I can tell, this >>>>>> company went bankrupt, and the ebay seller bought the phones from a >>>>>> bankruptcy auction. He does not know the admin username or password. >>>>>> Nowhere on the linksys site is there a solution to how to unlock these >>>>>> phones. >>>>>> >>>>>> Is there a way, or did I buy 2 interesting looking doorstops? Other >>>>>> than the password thing, they function fine. >>>>>> >>>>>> Thanks - >>>>>> >>>>>> Library Mark >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mattdfong at gmail.com Fri Mar 20 06:04:10 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 20 Mar 2009 20:04:10 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> Message-ID: <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> Hi Anthony, I'm trying to use fifo in a different sense. Instead of using it for inbound call queing, I'd like to use it for outbound call making. So instead, my agents are "waiting" in the que, and once an outbound call is connected, the "caller" will take an agent out of the que. So, in my case, the Fifo agent, would not be able to transfer the call because it's an outbound call, and the phone number on the other side is that of a non-employee. Fifo works a little smoother this way, because in reality, for outbound call making to an agent, this is what's happening, not vica versa. How difficult would this be to implement? Thanks. --matt 2009/3/20 Anthony Minessale > The agent could transfer the caller to another extension. > > > 2009/3/19 Matthew Fong > > Hi Anthony, >> I installed the patch, but I don't think it accomplishes what I want. >> >> I want the opposite, I want the fifo caller to continue along with the >> dialplan after the agent (consumer) is finished with servicing the call. >> This might be useful in situations where there could be an IVR recording >> customer satisfaction results of the agent servicing the call. As is, FS >> ends the caller's channel after finishing up in the fifo (ie, agent >> (consumer) disconnects or hangsup)--there should be life after s/he has been >> serviced by an agent (preferably continuing on in the dialplan). >> >> If I'm confused and missing something obvious, please correct me... Thanks >> >> --matt >> >> >> >> 2009/3/19 Anthony Minessale >> >> This is the patch >>> >>> http://jira.freeswitch.org/browse/MODAPP-237 >>> >>> it's not added yet. >>> >>> >>> 2009/3/18 Matthew Fong >>> >>> I upgraded to >>>> FreeSWITCH Version 1.0.trunk (12654M) >>>> >>>> but caller is still being hungup (and not continuing on with dialplan) >>>> after agent disconnect with hangup_after_bridge=false >>>> >>>> Is there a separate patch I need to apply? Thanks. >>>> >>>> --matt >>>> >>>> >>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>>> >>>>> Hi Anthony, thanks for the reply. >>>>> I've searched thru jira, and didn't find anything when searching for >>>>> fifo that was recently updated or related, except >>>>> >>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>> >>>>> and I'm not sure if this does what I need. Was this what you were >>>>> referring to? Thanks. >>>>> >>>>> --matt >>>>> >>>>> 2009/3/17 Anthony Minessale >>>>> >>>>> there is a patch in jira that will implement this feature about to be >>>>>> added >>>>>> >>>>>> >>>>>> 2009/3/17 Matthew Fong >>>>>> >>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>> (first timer posting here ;). >>>>>>> >>>>>>> I have a situation where it would be useful for a caller not to be >>>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>>> situation. It would be preferable if the caller channel went further along >>>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>> could be done as a feature enhancement? Thanks. >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/374b2ced/attachment-0001.html From mattdfong at gmail.com Fri Mar 20 06:06:35 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 20 Mar 2009 20:06:35 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> Message-ID: <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> Also, I would not be able to have a hang-up script do it, because in this scenario, the fifo consumer could hang-up at any time without any prior warning--otherwise, I could just transfer the fifo caller out before the fifo agent hangsup...but there is no prior warning :( --matt On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong wrote: > Hi Anthony, > I'm trying to use fifo in a different sense. Instead of using it for > inbound call queing, I'd like to use it for outbound call making. So > instead, my agents are "waiting" in the que, and once an outbound call is > connected, the "caller" will take an agent out of the que. > > So, in my case, the Fifo agent, would not be able to transfer the call > because it's an outbound call, and the phone number on the other side is > that of a non-employee. > > Fifo works a little smoother this way, because in reality, for outbound > call making to an agent, this is what's happening, not vica versa. How > difficult would this be to implement? Thanks. > > --matt > > 2009/3/20 Anthony Minessale > > The agent could transfer the caller to another extension. >> >> >> 2009/3/19 Matthew Fong >> >> Hi Anthony, >>> I installed the patch, but I don't think it accomplishes what I want. >>> >>> I want the opposite, I want the fifo caller to continue along with the >>> dialplan after the agent (consumer) is finished with servicing the call. >>> This might be useful in situations where there could be an IVR recording >>> customer satisfaction results of the agent servicing the call. As is, FS >>> ends the caller's channel after finishing up in the fifo (ie, agent >>> (consumer) disconnects or hangsup)--there should be life after s/he has been >>> serviced by an agent (preferably continuing on in the dialplan). >>> >>> If I'm confused and missing something obvious, please correct me... >>> Thanks >>> >>> --matt >>> >>> >>> >>> 2009/3/19 Anthony Minessale >>> >>> This is the patch >>>> >>>> http://jira.freeswitch.org/browse/MODAPP-237 >>>> >>>> it's not added yet. >>>> >>>> >>>> 2009/3/18 Matthew Fong >>>> >>>> I upgraded to >>>>> FreeSWITCH Version 1.0.trunk (12654M) >>>>> >>>>> but caller is still being hungup (and not continuing on with dialplan) >>>>> after agent disconnect with hangup_after_bridge=false >>>>> >>>>> Is there a separate patch I need to apply? Thanks. >>>>> >>>>> --matt >>>>> >>>>> >>>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>>>> >>>>>> Hi Anthony, thanks for the reply. >>>>>> I've searched thru jira, and didn't find anything when searching for >>>>>> fifo that was recently updated or related, except >>>>>> >>>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>>> >>>>>> and I'm not sure if this does what I need. Was this what you were >>>>>> referring to? Thanks. >>>>>> >>>>>> --matt >>>>>> >>>>>> 2009/3/17 Anthony Minessale >>>>>> >>>>>> there is a patch in jira that will implement this feature about to be >>>>>>> added >>>>>>> >>>>>>> >>>>>>> 2009/3/17 Matthew Fong >>>>>>> >>>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>>> (first timer posting here ;). >>>>>>>> >>>>>>>> I have a situation where it would be useful for a caller not to be >>>>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>>>> situation. It would be preferable if the caller channel went further along >>>>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>>> could be done as a feature enhancement? Thanks. >>>>>>>> >>>>>>>> --matt >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> >>>>>>> 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:213-799-1400 >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/a193b6b3/attachment.html From anthony.minessale at gmail.com Fri Mar 20 06:25:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Mar 2009 08:25:44 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> Message-ID: <191c3a030903200625k23edd429p38ec8b518ec97ee3@mail.gmail.com> Even though it's an outbound call if your agent uses his sip phone to blind transfer the caller "customer", The customer call will change the the routing state and hunt in your local dialplan just like it was an inbound call. That's how blind transfer was designed to work. If your agent is not using a sip phone, you can use bind_meta_app to make *N (where N = 0-9) to trigger a software blind transfer. 2009/3/20 Matthew Fong > Also, I would not be able to have a hang-up script do it, because in this > scenario, the fifo consumer could hang-up at any time without any prior > warning--otherwise, I could just transfer the fifo caller out before the > fifo agent hangsup...but there is no prior warning :( > --matt > > > On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong wrote: > >> Hi Anthony, >> I'm trying to use fifo in a different sense. Instead of using it for >> inbound call queing, I'd like to use it for outbound call making. So >> instead, my agents are "waiting" in the que, and once an outbound call is >> connected, the "caller" will take an agent out of the que. >> >> So, in my case, the Fifo agent, would not be able to transfer the call >> because it's an outbound call, and the phone number on the other side is >> that of a non-employee. >> >> Fifo works a little smoother this way, because in reality, for outbound >> call making to an agent, this is what's happening, not vica versa. How >> difficult would this be to implement? Thanks. >> >> --matt >> >> 2009/3/20 Anthony Minessale >> >> The agent could transfer the caller to another extension. >>> >>> >>> 2009/3/19 Matthew Fong >>> >>> Hi Anthony, >>>> I installed the patch, but I don't think it accomplishes what I want. >>>> >>>> I want the opposite, I want the fifo caller to continue along with the >>>> dialplan after the agent (consumer) is finished with servicing the call. >>>> This might be useful in situations where there could be an IVR recording >>>> customer satisfaction results of the agent servicing the call. As is, FS >>>> ends the caller's channel after finishing up in the fifo (ie, agent >>>> (consumer) disconnects or hangsup)--there should be life after s/he has been >>>> serviced by an agent (preferably continuing on in the dialplan). >>>> >>>> If I'm confused and missing something obvious, please correct me... >>>> Thanks >>>> >>>> --matt >>>> >>>> >>>> >>>> 2009/3/19 Anthony Minessale >>>> >>>> This is the patch >>>>> >>>>> http://jira.freeswitch.org/browse/MODAPP-237 >>>>> >>>>> it's not added yet. >>>>> >>>>> >>>>> 2009/3/18 Matthew Fong >>>>> >>>>> I upgraded to >>>>>> FreeSWITCH Version 1.0.trunk (12654M) >>>>>> >>>>>> but caller is still being hungup (and not continuing on with dialplan) >>>>>> after agent disconnect with hangup_after_bridge=false >>>>>> >>>>>> Is there a separate patch I need to apply? Thanks. >>>>>> >>>>>> --matt >>>>>> >>>>>> >>>>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>>>>> >>>>>>> Hi Anthony, thanks for the reply. >>>>>>> I've searched thru jira, and didn't find anything when searching for >>>>>>> fifo that was recently updated or related, except >>>>>>> >>>>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>>>> >>>>>>> and I'm not sure if this does what I need. Was this what you were >>>>>>> referring to? Thanks. >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> 2009/3/17 Anthony Minessale >>>>>>> >>>>>>> there is a patch in jira that will implement this feature about to be >>>>>>>> added >>>>>>>> >>>>>>>> >>>>>>>> 2009/3/17 Matthew Fong >>>>>>>> >>>>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>>>> (first timer posting here ;). >>>>>>>>> >>>>>>>>> I have a situation where it would be useful for a caller not to be >>>>>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>>>>> situation. It would be preferable if the caller channel went further along >>>>>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>>>> could be done as a feature enhancement? Thanks. >>>>>>>>> >>>>>>>>> --matt >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II >>>>>>>> >>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>> >>>>>>>> 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:213-799-1400 >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/0a3d4826/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 20 06:48:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Mar 2009 08:48:41 -0500 Subject: [Freeswitch-users] not hanging up In-Reply-To: <4ea6e8f20903191242vb56bcaetb7acd4853b215b0e@mail.gmail.com> References: <4ea6e8f20903191242vb56bcaetb7acd4853b215b0e@mail.gmail.com> Message-ID: <191c3a030903200648g9b10d78h5b81a79cbbeaa107@mail.gmail.com> It looks like interop issue with dialog matching between asterisk and freeswitch. Which version of asterisk is it? Which version of FreeSWITCH? You may want to provide a trace of the whole call starting with the invite. FS is having trouble identifying what call asterisk wants to cancel. 2009/3/19 Steven Ward > I have phones registered to a FS box, and an * box. There is a sip trunk > between the two boxes. > > A phone on my * (54321) calls a FS phone (12345); if I hang up the * phone > while it's still ringing, this is what I get on the sip trace on FS: > > ... > > 2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c:1692 > switch_ivr_originate() Ring Ready sofia/internal/12345 at 11.2.22.45! > recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950: > ------------------------------------------------------------------------ > CANCEL sip:12345 at b-pbx-sip-3.abc.xyz.netSIP/2.0 > Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport > From: "Steve" > >;tag=as25193d44 > To: > > > Call-ID: 0c0614d866a62841546cbf3340224682 at 11.2.22.45 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 328 bytes to udp/[11.2.22.45]:5060 at 19:05:44.313572: > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport=5060 > From: "Steve" > >;tag=as25193d44 > To: > >;tag=c5Z8Q1e93p7KD > Call-ID: 0c0614d866a62841546cbf3340224682 at 11.2.22.45 > CSeq: 103 CANCEL > Content-Length: 0 > > -------------------------------------------------------- > > > The effect is that the FS keeps on ringing - it doesn't detect the hangup. > > > When I call from a FS phone (1000) to another FS phone (12345), and I hang > up the calling phone > while it's still ringing, this is what I get on the sip trace: > > ... > > send 425 bytes to udp/[11.2.56.106]:63054 at 19:15:29.737163: > ------------------------------------------------------------------------ > CANCEL sip:12345 at 11.2.56.106:63054;rinstance=64e968d7a1317bc3 SIP/2.0 > Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a > Max-Forwards: 69 > From: "Extension 1000" > >;tag=meK8yUgpgU2Zc > To: > Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 > CSeq: 112626727 CANCEL > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 427 bytes from udp/[11.2.56.106]:63054 at 19:15:29.838863: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a > Contact: > To: ;rinstance=64e968d7a1317bc3>;tag=db12c87a > From: "Extension 1000" > >;tag=meK8yUgpgU2Zc > Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 > CSeq: 112626727 CANCEL > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 376 bytes from udp/[11.2.56.106]:63054 at 19:15:29.839334: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a > To: ;rinstance=64e968d7a1317bc3>;tag=db12c87a > From: "Extension 1000" > >;tag=meK8yUgpgU2Zc > Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 > CSeq: 112626727 INVITE > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > > ... > > It works just fine. Any ideas? I'm not sure where to go with this. > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/f2b9e550/attachment.html From mattdfong at gmail.com Fri Mar 20 07:27:31 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 20 Mar 2009 21:27:31 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <191c3a030903200625k23edd429p38ec8b518ec97ee3@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> <191c3a030903200625k23edd429p38ec8b518ec97ee3@mail.gmail.com> Message-ID: <4256bf830903200727g7259549dk8af17b4c583848d8@mail.gmail.com> Hi Anthony, Thanks for explaining blind transfer for me. The issue is that the fifo caller (my employee/agent, fifo in), gets hung-up on when the fifo consumer (an outside line to another party, fifo out) hangs up. I think this is because fifo was written under the assumption that the first in first out would always be a caller, and the agent would consume a caller. In my case, the roles are reversed, and there's no option to prevent the hangup of the caller. If the fifo caller (my employee/agent) could somehow know when a fifo consumer (my outside line to another party) was going to hangup, s/he could blind transfer out to save his/her connection from being hung-up, but unfortunately people don't always tell you before hand they are going to hangup. Right?!?!?! Thanks. --matt 2009/3/20 Anthony Minessale > Even though it's an outbound call if your agent uses his sip phone to blind > transfer the caller "customer", > The customer call will change the the routing state and hunt in your local > dialplan just like it was an inbound call. That's how blind transfer was > designed to work. > > If your agent is not using a sip phone, you can use bind_meta_app to make > *N (where N = 0-9) to trigger a software blind transfer. > > > 2009/3/20 Matthew Fong > > Also, I would not be able to have a hang-up script do it, because in this >> scenario, the fifo consumer could hang-up at any time without any prior >> warning--otherwise, I could just transfer the fifo caller out before the >> fifo agent hangsup...but there is no prior warning :( >> --matt >> >> >> On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong wrote: >> >>> Hi Anthony, >>> I'm trying to use fifo in a different sense. Instead of using it for >>> inbound call queing, I'd like to use it for outbound call making. So >>> instead, my agents are "waiting" in the que, and once an outbound call is >>> connected, the "caller" will take an agent out of the que. >>> >>> So, in my case, the Fifo agent, would not be able to transfer the call >>> because it's an outbound call, and the phone number on the other side is >>> that of a non-employee. >>> >>> Fifo works a little smoother this way, because in reality, for outbound >>> call making to an agent, this is what's happening, not vica versa. How >>> difficult would this be to implement? Thanks. >>> >>> --matt >>> >>> 2009/3/20 Anthony Minessale >>> >>> The agent could transfer the caller to another extension. >>>> >>>> >>>> 2009/3/19 Matthew Fong >>>> >>>> Hi Anthony, >>>>> I installed the patch, but I don't think it accomplishes what I want. >>>>> >>>>> I want the opposite, I want the fifo caller to continue along with the >>>>> dialplan after the agent (consumer) is finished with servicing the call. >>>>> This might be useful in situations where there could be an IVR recording >>>>> customer satisfaction results of the agent servicing the call. As is, FS >>>>> ends the caller's channel after finishing up in the fifo (ie, agent >>>>> (consumer) disconnects or hangsup)--there should be life after s/he has been >>>>> serviced by an agent (preferably continuing on in the dialplan). >>>>> >>>>> If I'm confused and missing something obvious, please correct me... >>>>> Thanks >>>>> >>>>> --matt >>>>> >>>>> >>>>> >>>>> 2009/3/19 Anthony Minessale >>>>> >>>>> This is the patch >>>>>> >>>>>> http://jira.freeswitch.org/browse/MODAPP-237 >>>>>> >>>>>> it's not added yet. >>>>>> >>>>>> >>>>>> 2009/3/18 Matthew Fong >>>>>> >>>>>> I upgraded to >>>>>>> FreeSWITCH Version 1.0.trunk (12654M) >>>>>>> >>>>>>> but caller is still being hungup (and not continuing on with >>>>>>> dialplan) after agent disconnect with hangup_after_bridge=false >>>>>>> >>>>>>> Is there a separate patch I need to apply? Thanks. >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> >>>>>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>>>>>> >>>>>>>> Hi Anthony, thanks for the reply. >>>>>>>> I've searched thru jira, and didn't find anything when searching for >>>>>>>> fifo that was recently updated or related, except >>>>>>>> >>>>>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>>>>> >>>>>>>> and I'm not sure if this does what I need. Was this what you were >>>>>>>> referring to? Thanks. >>>>>>>> >>>>>>>> --matt >>>>>>>> >>>>>>>> 2009/3/17 Anthony Minessale >>>>>>>> >>>>>>>> there is a patch in jira that will implement this feature about to >>>>>>>>> be added >>>>>>>>> >>>>>>>>> >>>>>>>>> 2009/3/17 Matthew Fong >>>>>>>>> >>>>>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>>>>> (first timer posting here ;). >>>>>>>>>> >>>>>>>>>> I have a situation where it would be useful for a caller not to be >>>>>>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>>>>>> situation. It would be preferable if the caller channel went further along >>>>>>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>>>>> could be done as a feature enhancement? Thanks. >>>>>>>>>> >>>>>>>>>> --matt >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Freeswitch-users mailing list >>>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Anthony Minessale II >>>>>>>>> >>>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>>> >>>>>>>>> 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:213-799-1400 >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/7a982ed3/attachment-0001.html From Mark.Tabron at rnid-typetalk.org.uk Fri Mar 20 09:09:00 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Fri, 20 Mar 2009 16:09:00 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL><49C12CD8.7020203@gmx.net><11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903190911s6082877at1231c27f6a86506@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765381@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765382@tt-mail.RNID.TYPETALK.LOCAL> Installed libpri but I'm stuck on what entries to put in openzap.conf.xml, here's how I have the span setup at the moment: Node and Switch type aren't documented for libpri from what I can tell - I know the former is either CPE or NET, though, I'm unsure what other values can be used for switch type. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 19 March 2009 16:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Thanks, found an install guide on the FS Wiki for libpri - will get the server cloned then install and test. Shall report back. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 19 March 2009 16:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron wrote: > So the second issue is possibly known - really could do with a fix or a > workaround for this as we plan to use E1's for all incoming traffic. > > Can anyone shed light on the first problem (extension rings for a > fraction of a second then hangs up) I mentioned below, or is that > possibly part of the same issue? I have experienced this before but I believe it was resolved by having the telco switch protocol dialects which is probably not an option for you. I think your best bet is to use ozmod_libpri and see if the issue is still present. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. ------------------------------------------------------------------------ -------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). ------------------------------------------------------------------------ -------- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From stkn at freeswitch.org Fri Mar 20 11:29:05 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Fri, 20 Mar 2009 19:29:05 +0100 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE765382@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765381@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765382@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <200903201929.05829.stkn@freeswitch.org> Am Friday 20 March 2009 schrieb Mark Tabron: > Installed libpri but I'm stuck on what entries to put in > openzap.conf.xml, here's how I have the span setup at the moment: > > > > > > > > > > > Node and Switch type aren't documented for libpri from what I can tell - > I know the former is either CPE or NET, though, I'm unsure what other > values can be used for switch type. > The value for switch is invalid, it's going to fall back to dms100 with that one set. Valid settings are: ni1, ni2, dms100, euroisdn, lucent5e, att4ess, gr303eoc and gr303tmc and "euroisdn" is the one you'll want for a E1 line. Another setting you may need is: stkn -- ------------------------------------------------------------------------------- Stefan Knoblich | axsentis GmbH | Web: http://www.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | http://oss.axsentis.de/ Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From anthony.minessale at gmail.com Fri Mar 20 12:04:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Mar 2009 14:04:04 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903200727g7259549dk8af17b4c583848d8@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> <191c3a030903200625k23edd429p38ec8b518ec97ee3@mail.gmail.com> <4256bf830903200727g7259549dk8af17b4c583848d8@mail.gmail.com> Message-ID: <191c3a030903201204l5dc16ad1pee4f4635e1addef7@mail.gmail.com> I added a feature to latest trunk where you can set the variable transfer_after_bridge=xyz where xyz is an extension you want to transfer to when the call ends assuming it's not hungup yet. if you set this in your dialplan before entering fifo it will allow you to be transfered to the desired ext when you are done. if you need to specify a dialplan and context you cana add that too [:][:] 2009/3/20 Matthew Fong > Hi Anthony, > Thanks for explaining blind transfer for me. > > The issue is that the fifo caller (my employee/agent, fifo in), gets > hung-up on when the fifo consumer (an outside line to another party, fifo > out) hangs up. I think this is because fifo was written under the assumption > that the first in first out would always be a caller, and the agent would > consume a caller. > > In my case, the roles are reversed, and there's no option to prevent the > hangup of the caller. > > If the fifo caller (my employee/agent) could somehow know when a fifo > consumer (my outside line to another party) was going to hangup, s/he could > blind transfer out to save his/her connection from being hung-up, but > unfortunately people don't always tell you before hand they are going to > hangup. > > Right?!?!?! Thanks. > > --matt > > 2009/3/20 Anthony Minessale > >> Even though it's an outbound call if your agent uses his sip phone to >> blind transfer the caller "customer", >> The customer call will change the the routing state and hunt in your local >> dialplan just like it was an inbound call. That's how blind transfer was >> designed to work. >> >> If your agent is not using a sip phone, you can use bind_meta_app to make >> *N (where N = 0-9) to trigger a software blind transfer. >> >> >> 2009/3/20 Matthew Fong >> >> Also, I would not be able to have a hang-up script do it, because in this >>> scenario, the fifo consumer could hang-up at any time without any prior >>> warning--otherwise, I could just transfer the fifo caller out before the >>> fifo agent hangsup...but there is no prior warning :( >>> --matt >>> >>> >>> On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong wrote: >>> >>>> Hi Anthony, >>>> I'm trying to use fifo in a different sense. Instead of using it for >>>> inbound call queing, I'd like to use it for outbound call making. So >>>> instead, my agents are "waiting" in the que, and once an outbound call is >>>> connected, the "caller" will take an agent out of the que. >>>> >>>> So, in my case, the Fifo agent, would not be able to transfer the call >>>> because it's an outbound call, and the phone number on the other side is >>>> that of a non-employee. >>>> >>>> Fifo works a little smoother this way, because in reality, for outbound >>>> call making to an agent, this is what's happening, not vica versa. How >>>> difficult would this be to implement? Thanks. >>>> >>>> --matt >>>> >>>> 2009/3/20 Anthony Minessale >>>> >>>> The agent could transfer the caller to another extension. >>>>> >>>>> >>>>> 2009/3/19 Matthew Fong >>>>> >>>>> Hi Anthony, >>>>>> I installed the patch, but I don't think it accomplishes what I want. >>>>>> >>>>>> I want the opposite, I want the fifo caller to continue along with the >>>>>> dialplan after the agent (consumer) is finished with servicing the call. >>>>>> This might be useful in situations where there could be an IVR recording >>>>>> customer satisfaction results of the agent servicing the call. As is, FS >>>>>> ends the caller's channel after finishing up in the fifo (ie, agent >>>>>> (consumer) disconnects or hangsup)--there should be life after s/he has been >>>>>> serviced by an agent (preferably continuing on in the dialplan). >>>>>> >>>>>> If I'm confused and missing something obvious, please correct me... >>>>>> Thanks >>>>>> >>>>>> --matt >>>>>> >>>>>> >>>>>> >>>>>> 2009/3/19 Anthony Minessale >>>>>> >>>>>> This is the patch >>>>>>> >>>>>>> http://jira.freeswitch.org/browse/MODAPP-237 >>>>>>> >>>>>>> it's not added yet. >>>>>>> >>>>>>> >>>>>>> 2009/3/18 Matthew Fong >>>>>>> >>>>>>> I upgraded to >>>>>>>> FreeSWITCH Version 1.0.trunk (12654M) >>>>>>>> >>>>>>>> but caller is still being hungup (and not continuing on with >>>>>>>> dialplan) after agent disconnect with hangup_after_bridge=false >>>>>>>> >>>>>>>> Is there a separate patch I need to apply? Thanks. >>>>>>>> >>>>>>>> --matt >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong >>>>>>> > wrote: >>>>>>>> >>>>>>>>> Hi Anthony, thanks for the reply. >>>>>>>>> I've searched thru jira, and didn't find anything when searching >>>>>>>>> for fifo that was recently updated or related, except >>>>>>>>> >>>>>>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>>>>>> >>>>>>>>> and I'm not sure if this does what I need. Was this what you were >>>>>>>>> referring to? Thanks. >>>>>>>>> >>>>>>>>> --matt >>>>>>>>> >>>>>>>>> 2009/3/17 Anthony Minessale >>>>>>>>> >>>>>>>>> there is a patch in jira that will implement this feature about to >>>>>>>>>> be added >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2009/3/17 Matthew Fong >>>>>>>>>> >>>>>>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>>>>>> (first timer posting here ;). >>>>>>>>>>> >>>>>>>>>>> I have a situation where it would be useful for a caller not to >>>>>>>>>>> be hungup, after finishing the "fifo in" execution (when the agent >>>>>>>>>>> disconnects the call or the agent hangs-up). The caller is automatically >>>>>>>>>>> hungup, in this situation. It would be preferable if the caller channel went >>>>>>>>>>> further along the dial plan. I thought I might get lucky implementing this >>>>>>>>>>> setting with hangup_after_bridge to false, but fifo does not utilize this >>>>>>>>>>> variable. >>>>>>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>>>>>> could be done as a feature enhancement? Thanks. >>>>>>>>>>> >>>>>>>>>>> --matt >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> Freeswitch-users mailing list >>>>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Anthony Minessale II >>>>>>>>>> >>>>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>>>> >>>>>>>>>> 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:213-799-1400 >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Freeswitch-users mailing list >>>>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> >>>>>>> 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:213-799-1400 >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/7f6214fa/attachment-0001.html From switchserver at gmail.com Fri Mar 20 13:14:02 2009 From: switchserver at gmail.com (HarryK) Date: Fri, 20 Mar 2009 13:14:02 -0700 (PDT) Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> Message-ID: <22628039.post@talk.nabble.com> Yes that page was perfect, ty. (Cepstral 5.1) One thing. Is there a way to set the default voice used? I have 2 voices installed and it goes to the first one by default. It also refers to this voice as "David" even though its Allison. I'd like it to always use the second voice for RSS. There is nothing in the configs, I figured maybe it was yet to be documented. I hope Cepstral is more stable in the future, I havent used it much so noticed no problems yet. Thanks again Carlos Talbot wrote: > > I wrote this wiki page a while back. Did it help? > > http://wiki.freeswitch.org/wiki/Mod_rss > > > On Thu, Mar 19, 2009 at 2:41 AM, HarryK wrote: > >> >> I have Cepstral working. >> >> Can someone please tell me how to go about having it read RSS feeds? I >> can >> have the dialplan direct it np. But I really dont have a clue how to >> point >> it at an RSS. Any help would be great, ddint find anything in the wiki. >> >> >> -- >> View this message in context: >> http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22594923.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22628039.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gkuri at ieee.org Fri Mar 20 14:28:35 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 20 Mar 2009 14:28:35 -0700 Subject: [Freeswitch-users] PCMU fallback for T.38 Message-ID: <49C40A83.1050003@ieee.org> hey folks, I'm trying to configure PCMU fallback for T.38. The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with G729 and PCMU in the sdp. the INVITE to the provider includes G729 and PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... m=audio 16458 RTP/AVP 18 0 100 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 once the FAX tone is detected on the PSTN side, FS receives a T.38 re-INVITE from the provider and FS sends back a 488/Not Acceptable (proxy_media=false). at that point the provider than attempts fallback to PCMU with another reINVITE ... m=audio 16816 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 immediately after the PCMU reINVITE, FS closes the channel and the text below is in the FS logs. given the SPA-2102 included PCMU in the original INVITE, even though it was the second preferred codec, shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by the provider? It seems like it's not passing the PCMU Re-INVITE back to the endpoint (SPA-2102), since it originally negotiated G729 with the SPA2102 as that was the 1st codec in the sdp, but trying to transcode between the two (G729 and PCMU)? 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 PCMU/8000 20 ms 160 samples 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1. 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing Reinvite 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 entering state [completed] 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! Thanks, Gabe From brian at freeswitch.org Fri Mar 20 14:35:14 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Mar 2009 16:35:14 -0500 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49C40A83.1050003@ieee.org> References: <49C40A83.1050003@ieee.org> Message-ID: <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> Are you on SVN trunk 12694? /b On Mar 20, 2009, at 4:28 PM, Gabriel Kuri wrote: > hey folks, I'm trying to configure PCMU fallback for T.38. > > The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with > G729 and PCMU in the sdp. the INVITE to the provider includes G729 and > PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... > > m=audio 16458 RTP/AVP 18 0 100 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > once the FAX tone is detected on the PSTN side, FS receives a T.38 > re-INVITE from the provider and FS sends back a 488/Not Acceptable > (proxy_media=false). at that point the provider than attempts fallback > to PCMU with another reINVITE ... > > m=audio 16816 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > immediately after the PCMU reINVITE, FS closes the channel and the > text > below is in the FS logs. given the SPA-2102 included PCMU in the > original INVITE, even though it was the second preferred codec, > shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by > the provider? It seems like it's not passing the PCMU Re-INVITE back > to > the endpoint (SPA-2102), since it originally negotiated G729 with the > SPA2102 as that was the 1st codec in the sdp, but trying to transcode > between the two (G729 and PCMU)? > > > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 > sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 > sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 > sofia_glue_negotiate_sdp() > Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 > sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 > sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 > sofia_glue_tech_set_codec() Set Codec > sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 PCMU/8000 20 ms 160 > samples > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 > sofia_glue_activate_rtp() > Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 > . > 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing Reinvite > 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 entering state > [completed] > 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 > switch_core_session_write_frame() > sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 receive message > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This > codec > is only usable in passthrough mode! > 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 > switch_core_session_write_frame() Codec G.729 decoder error! > > > Thanks, > > Gabe > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gkuri at ieee.org Fri Mar 20 14:41:47 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 20 Mar 2009 14:41:47 -0700 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> References: <49C40A83.1050003@ieee.org> <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> Message-ID: <49C40D9B.6000803@ieee.org> err, no, I tried upgrading from r11000 to r12669 yesterday, but starting seeing crashing, so I have a jira open. currently I'm back on r11000. http://jira.freeswitch.org/browse/FSCORE-338 Gabe Brian West wrote: > Are you on SVN trunk 12694? > > /b > > On Mar 20, 2009, at 4:28 PM, Gabriel Kuri wrote: > >> hey folks, I'm trying to configure PCMU fallback for T.38. >> >> The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with >> G729 and PCMU in the sdp. the INVITE to the provider includes G729 and >> PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... >> >> m=audio 16458 RTP/AVP 18 0 100 101 >> a=rtpmap:18 G729/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:100 NSE/8000 >> a=fmtp:100 192-193 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> once the FAX tone is detected on the PSTN side, FS receives a T.38 >> re-INVITE from the provider and FS sends back a 488/Not Acceptable >> (proxy_media=false). at that point the provider than attempts fallback >> to PCMU with another reINVITE ... >> >> m=audio 16816 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> immediately after the PCMU reINVITE, FS closes the channel and the >> text >> below is in the FS logs. given the SPA-2102 included PCMU in the >> original INVITE, even though it was the second preferred codec, >> shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by >> the provider? It seems like it's not passing the PCMU Re-INVITE back >> to >> the endpoint (SPA-2102), since it originally negotiated G729 with the >> SPA2102 as that was the 1st codec in the sdp, but trying to transcode >> between the two (G729 and PCMU)? >> >> >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 >> sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 >> sofia_glue_negotiate_sdp() >> Set 2833 dtmf payload to 101 >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 >> sofia_glue_negotiate_sdp() >> Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 >> sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 >> sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 >> sofia_glue_tech_set_codec() Set Codec >> sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 PCMU/8000 20 ms 160 >> samples >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 >> sofia_glue_activate_rtp() >> Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 >> . >> 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() >> Processing Reinvite >> 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() >> Channel sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 entering state >> [completed] >> 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 >> switch_core_session_write_frame() >> sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 receive message >> [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] >> 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This >> codec >> is only usable in passthrough mode! >> 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 >> switch_core_session_write_frame() Codec G.729 decoder error! >> >> >> Thanks, >> >> Gabe >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri Mar 20 14:48:43 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Mar 2009 16:48:43 -0500 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49C40D9B.6000803@ieee.org> References: <49C40A83.1050003@ieee.org> <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> <49C40D9B.6000803@ieee.org> Message-ID: <59121C75-9F16-4794-AA91-9F71DA3B6E7D@freeswitch.org> Make current and try again... I haven't seen this crash you have seen... if you can run sippcapdump and get the packets that would help also. Thanks, /b On Mar 20, 2009, at 4:41 PM, Gabriel Kuri wrote: > err, no, I tried upgrading from r11000 to r12669 yesterday, but > starting > seeing crashing, so I have a jira open. currently I'm back on r11000. > > http://jira.freeswitch.org/browse/FSCORE-338 > > Gabe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/814dd93d/attachment.html From gkuri at ieee.org Fri Mar 20 16:04:07 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 20 Mar 2009 16:04:07 -0700 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <59121C75-9F16-4794-AA91-9F71DA3B6E7D@freeswitch.org> References: <49C40A83.1050003@ieee.org> <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> <49C40D9B.6000803@ieee.org> <59121C75-9F16-4794-AA91-9F71DA3B6E7D@freeswitch.org> Message-ID: <49C420E7.0@ieee.org> OK, I'll give it a try and report back. Gabe Brian West wrote: > Make current and try again... I haven't seen this crash you have seen... > if you can run sippcapdump and get the packets that would help also. > > Thanks, > /b > > > On Mar 20, 2009, at 4:41 PM, Gabriel Kuri wrote: > >> err, no, I tried upgrading from r11000 to r12669 yesterday, but starting >> seeing crashing, so I have a jira open. currently I'm back on r11000. >> >> http://jira.freeswitch.org/browse/FSCORE-338 >> >> Gabe > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave at 3c.co.uk Fri Mar 20 18:53:46 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 20 Mar 2009 19:53:46 -0600 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> Message-ID: <49C448AA.6070207@3c.co.uk> In the meantime, you can work around this by using the swift executable to turn text in to WAV files, and then just play them back. Works fine for short(ish) texts - there might be a bit of a delay if you wanted the thing to read back War and Peace. --Dave > http://jira.freeswitch.org/browse/MODASRTTS-11 > > Might wanna know about that issue also :) > > /b > > On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote: > >> I wrote this wiki page a while back. Did it help? >> >> http://wiki.freeswitch.org/wiki/Mod_rss > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/25851bbb/attachment-0001.html From brian at freeswitch.org Fri Mar 20 19:02:40 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Mar 2009 21:02:40 -0500 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <49C448AA.6070207@3c.co.uk> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> <49C448AA.6070207@3c.co.uk> Message-ID: <1C27B16A-898C-4DAA-9B1C-3E7DB4A15F22@freeswitch.org> Oxymoron!?!?! :) /b On Mar 20, 2009, at 8:53 PM, David Knell wrote: > War and Peace. From jason at jasonjgw.net Fri Mar 20 19:10:28 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 21 Mar 2009 13:10:28 +1100 Subject: [Freeswitch-users] proper way to clean source tree after build? Message-ID: <20090321021028.GA5399@jdc.jasonjgw.net> I'm building under Debian. To clean the source tree after building, I've tried make clean debuild clean but what often happens is that after updating to a later svn revision and then trying to build, I get various errors during the compilation process that disappear if I make a clean checkout from the repository and run the build again. I have also found that using svn export to reproduce the tree in another directory, then building from that directory, doesn't always solve the problem. Is there a better way to clean up the source tree after updating it, to avoid these issues? I assume there are extraneous modifications or files that simply aren't being eliminated by the "clean" makefile targets. From dave at 3c.co.uk Fri Mar 20 19:34:12 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 20 Mar 2009 20:34:12 -0600 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <1C27B16A-898C-4DAA-9B1C-3E7DB4A15F22@freeswitch.org> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> <49C448AA.6070207@3c.co.uk> <1C27B16A-898C-4DAA-9B1C-3E7DB4A15F22@freeswitch.org> Message-ID: <49C45224.6010106@3c.co.uk> It's a book, Brian, and a long one at that, often printed in black and white - oops, there I go again ;-) I once referred to "oxymoronic hip-hop culture", only to be firmly told by a listener that there was nothing moronic about hip-hop.. Cheers -- Dave > Oxymoron!?!?! :) > > /b > > On Mar 20, 2009, at 8:53 PM, David Knell wrote: > > >> War and Peace. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/ed3c15b8/attachment.html From steveu at coppice.org Fri Mar 20 19:46:32 2009 From: steveu at coppice.org (Steve Underwood) Date: Sat, 21 Mar 2009 10:46:32 +0800 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49C40A83.1050003@ieee.org> References: <49C40A83.1050003@ieee.org> Message-ID: <49C45508.402@coppice.org> Gabriel Kuri wrote: > once the FAX tone is detected on the PSTN side, FS receives a T.38 > re-INVITE from the provider and FS sends back a 488/Not Acceptable > (proxy_media=false). at that point the provider than attempts fallback > to PCMU with another reINVITE ... > This part is interesting, and the subject of a discussion we had recently. A number of systems try that second re-invite after a 488, but the SIP specs say the call is pretty much dead after the 488 message is exchanged. Are they just hoping that maybe the other end will be non-compliant enough to keep the call alive, and recover its media mode, or haven't they read the specs? Steve From pablosaro at gmail.com Fri Mar 20 19:54:03 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 20 Mar 2009 23:54:03 -0300 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <153785972B6D42C99E8CAF7DE2F1A145@wsandy> References: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> <153785972B6D42C99E8CAF7DE2F1A145@wsandy> Message-ID: <247f8100903201954gcc81d83lf199c6a9b26463b4@mail.gmail.com> Hi Andy, Did you get Matheu's? > if you are behind NAT it is possible that your router "forgot" the mapping betweeen FS and your > provider, try adding to your gateway. Pablo 2009/3/20 Andy Ayers : > Thanks Brian, > > I've upgraded to 1.0.3 and things seem a little better but I'm still loosing > the gateway connection intermittently. I rebuilt the config on upgrade, is > there any possibility I've missed something? Is there a keep-alive setting > for a gateway or a re-connect after x or something. > > Many thanks for your help. > Andy > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian > West > Sent: 18 March 2009 14:08 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Losing Gateway registration > > Upgrade to 1.03 or SVN Trunk > /b > On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: > > Hi, > > I've recently ugrade to version 1.02 of freeswitch and am having some > problems with my gateway registrations. The gateway successfully registers > with my voip provider when freeswitch first starts but if left running it > seems to loose it's connection to my voip provider. I can get it to > reconnect with a sofia restart. I'm using the same provider and user account > as with the old version of the software. Can you suggest any reaosn why this > may be happening and how I can prevent it? > > Many thanks > Andy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From pablosaro at gmail.com Fri Mar 20 20:15:58 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Sat, 21 Mar 2009 00:15:58 -0300 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C34E54.9000000@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> <49C34E54.9000000@czerpak.eu> Message-ID: <247f8100903202015g6befe872v73fe4a53585d6a60@mail.gmail.com> I am not a Cisco expert, but as far as I know packetization period is configurable in Cisco. When you specify the codec for a dial peer, you can set up the ptime value in milliseconds. This is an optional argument in the Cisco command line and what happens when providers don't specify it is that your end assumes a convenient or default ptime value, that probably don't match with far end default... A solution would be to figure out what is the convenient value for an individual provider and set up FS to match it, or suggest your provider to specify a ptime in the first request. By the way, Cisco supports 10, 20, 30, 40, 50 and 60 as ptime values for codecs G.729, G.729A, G.729B and G.729AB. Pablo 2009/3/20 ?ukasz Czerpak : > Brian West pisze: >> Well you can't have ptime 60 one way and 20 the other it just won't >> work. ?Also I can't even think that this illegal codec was even tested >> at 60ms... Try it with ulaw and see what it does. ?or only allow >> G729 at 60i and see what it does. >> > > I made some tests with ulaw with success. There is no problem with ptime > negotiation. > > Will the g729 codec be fully (not passthrough) supported in FreeSWITCH? > > regards, > > -- > ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mszlazak at aol.com Fri Mar 20 23:22:10 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 21 Mar 2009 02:22:10 -0400 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <49C448AA.6070207@3c.co.uk> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> <49C448AA.6070207@3c.co.uk> Message-ID: <8CB7804802401C1-1388-2309@webmail-mf01.sysops.aol.com> This product has much better sounding TTS than Cepstral: http://www.neospeech.com/Default.aspx Maybe you can use Dave's suggestion and make WAV recordings from their demo text input box and then just clip off the initial portion. Mark. -----Original Message----- From: David Knell To: freeswitch-users at lists.freeswitch.org Sent: Fri, 20 Mar 2009 6:53 pm Subject: Re: [Freeswitch-users] Cepstral and RSS feeds In the meantime, you can work around this by using the swift executable to turn text in to WAV files, and then just play them back.? Works fine for short(ish) texts - there might be a bit of a delay if you wanted the thing to read back War and Peace. --Dave http://jira.freeswitch.org/browse/MODASRTTS-11 Might wanna know about that issue also :) /b On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote: I wrote this wiki page a while back. Did it help? http://wiki.freeswitch.org/wiki/Mod_rss _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090321/45f047da/attachment.html From anthony.minessale at gmail.com Sat Mar 21 05:56:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Mar 2009 07:56:02 -0500 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <8CB7804802401C1-1388-2309@webmail-mf01.sysops.aol.com> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> <49C448AA.6070207@3c.co.uk> <8CB7804802401C1-1388-2309@webmail-mf01.sysops.aol.com> Message-ID: <191c3a030903210556j3611b0edh172a83ed0789acd1@mail.gmail.com> the rss app takes a space sep string of 3 optional args to control the voice 2009/3/21 > This product has much better sounding TTS than Cepstral: > > http://www.neospeech.com/Default.aspx > > Maybe you can use Dave's suggestion and make WAV recordings from their demo > text input box and then just clip off the initial portion. > > Mark. > > > -----Original Message----- > From: David Knell > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, 20 Mar 2009 6:53 pm > Subject: Re: [Freeswitch-users] Cepstral and RSS feeds > > In the meantime, you can work around this by using the swift executable > to turn text in to WAV files, and then just play them back. Works fine for > short(ish) texts - there might be a bit of a delay if you wanted the thing > to > read back War and Peace. > > --Dave > > http://jira.freeswitch.org/browse/MODASRTTS-11 > Might wanna know about that issue also :) > > /b > > On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote: > > I wrote this wiki page a while back. Did it help? > > http://wiki.freeswitch.org/wiki/Mod_rss > > > ------------------------------ > > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > *A Bad Credit Score is 600 or Below. See yours in just 2 easy steps! > * > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090321/09228fd0/attachment-0001.html From alhakeem at gmail.com Sat Mar 21 11:46:54 2009 From: alhakeem at gmail.com (Abdul Hakeem) Date: Sat, 21 Mar 2009 18:46:54 -0000 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <247f8100903202015g6befe872v73fe4a53585d6a60@mail.gmail.com> References: <49C28D4F.4040307@czerpak.eu><2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com><49C29AF2.2000703@czerpak.eu><49C34E54.9000000@czerpak.eu> <247f8100903202015g6befe872v73fe4a53585d6a60@mail.gmail.com> Message-ID: The pvalue of the originator overrides whatever you might have configured on Cisco GW. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Pablo Hernan Saro Sent: 21 March 2009 03:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] ptime problem with provider (Cisco hardware) I am not a Cisco expert, but as far as I know packetization period is configurable in Cisco. When you specify the codec for a dial peer, you can set up the ptime value in milliseconds. This is an optional argument in the Cisco command line and what happens when providers don't specify it is that your end assumes a convenient or default ptime value, that probably don't match with far end default... A solution would be to figure out what is the convenient value for an individual provider and set up FS to match it, or suggest your provider to specify a ptime in the first request. By the way, Cisco supports 10, 20, 30, 40, 50 and 60 as ptime values for codecs G.729, G.729A, G.729B and G.729AB. Pablo 2009/3/20 ?ukasz Czerpak : > Brian West pisze: >> Well you can't have ptime 60 one way and 20 the other it just won't >> work. ?Also I can't even think that this illegal codec was even >> tested at 60ms... Try it with ulaw and see what it does. ?or only >> allow G729 at 60i and see what it does. >> > > I made some tests with ulaw with success. There is no problem with > ptime negotiation. > > Will the g729 codec be fully (not passthrough) supported in FreeSWITCH? > > regards, > > -- > ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From zhaoxxqq at 163.com Sun Mar 22 03:22:19 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Sun, 22 Mar 2009 18:22:19 +0800 Subject: [Freeswitch-users] Problem about always Double accept DTMFs Message-ID: <200903221822142515830@163.com> I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS to PSTN with DID numbers. For inband I connect it to FS?s demo_ivr. When I press the key, the FS accept always DOUBLE of key. The debug information like below. 2009-03-22 17:50:26 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 2009-03-22 17:50:26 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-03-22 17:50:26 [DEBUG] switch_ivr_menu.c:308 play_and_collect() waiting for 3/4 digits t/o 2000 2009-03-22 17:50:26 [DEBUG] sofia.c:3753 sofia_handle_sip_i_info() INFO DTMF(1) 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:353 play_and_collect() digits '11' 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:523 switch_ivr_menu_execute() IVR menu 'jtq_greating' caught invalid input '11' 2009-03-22 17:50:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-03-22 17:50:28 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/external/13323015 Can any friend can help me? Zhao Xiaoqiang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090322/5c47549a/attachment.html From dave at 3c.co.uk Sun Mar 22 06:02:56 2009 From: dave at 3c.co.uk (David Knell) Date: Sun, 22 Mar 2009 07:02:56 -0600 Subject: [Freeswitch-users] Problem about always Double accept DTMFs In-Reply-To: <200903221822142515830@163.com> References: <200903221822142515830@163.com> Message-ID: <49C63700.8010104@3c.co.uk> Hi - It looks like you're getting digits both in the RTP stream and as SIP INFO. Try adding to the SIP profile you're using for inbound calls. --Dave > I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS > to PSTN with DID numbers. For inband I connect it to FS's demo_ivr. > When I press the key, the FS accept always DOUBLE of key. The debug > information like below. > > 2009-03-22 17:50:26 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 > 2009-03-22 17:50:26 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file > 2009-03-22 17:50:26 [DEBUG] switch_ivr_menu.c:308 play_and_collect() waiting for 3/4 digits t/o 2000 > 2009-03-22 17:50:26 [DEBUG] sofia.c:3753 sofia_handle_sip_i_info() INFO DTMF(1) > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:353 play_and_collect() digits '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:523 switch_ivr_menu_execute() IVR menu 'jtq_greating' caught invalid input '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-03-22 17:50:28 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/external/13323015 > > > Can any friend can help me? > > Zhao Xiaoqiang > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090322/45822ad7/attachment.html From herman.griffin at gmail.com Fri Mar 20 22:33:34 2009 From: herman.griffin at gmail.com (Herman Griffin) Date: Fri, 20 Mar 2009 22:33:34 -0700 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <247f8100903201954gcc81d83lf199c6a9b26463b4@mail.gmail.com> References: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> <153785972B6D42C99E8CAF7DE2F1A145@wsandy> <247f8100903201954gcc81d83lf199c6a9b26463b4@mail.gmail.com> Message-ID: <4d6f26b0903202233v72046523s19b2dace2c8cce55@mail.gmail.com> I was having the same problem (i think). It resolved it by adding gateways to the /usr/local/freeswitch/conf/directory/default.xml file instead of the /usr/local/freeswitch/sip_profiles/external/ directory. It may have been a total coincidence because I was so frustrated with the problem that I start from scratch. I backed up my conf directory and started configuring based on the example.com.xml template. However, it worked. I have had a loss of registration problem since then. ./herman griffin On Fri, Mar 20, 2009 at 7:54 PM, Pablo Hernan Saro wrote: > Hi Andy, > > Did you get Matheu's? > > > if you are behind NAT it is possible that your router "forgot" the > mapping betweeen FS and your > provider, try adding value="30" /> to your gateway. > > Pablo > > 2009/3/20 Andy Ayers : > > Thanks Brian, > > > > I've upgraded to 1.0.3 and things seem a little better but I'm still > loosing > > the gateway connection intermittently. I rebuilt the config on upgrade, > is > > there any possibility I've missed something? Is there a keep-alive > setting > > for a gateway or a re-connect after x or something. > > > > Many thanks for your help. > > Andy > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian > > West > > Sent: 18 March 2009 14:08 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Losing Gateway registration > > > > Upgrade to 1.03 or SVN Trunk > > /b > > On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: > > > > Hi, > > > > I've recently ugrade to version 1.02 of freeswitch and am having some > > problems with my gateway registrations. The gateway successfully > registers > > with my voip provider when freeswitch first starts but if left running it > > seems to loose it's connection to my voip provider. I can get it to > > reconnect with a sofia restart. I'm using the same provider and user > account > > as with the old version of the software. Can you suggest any reaosn why > this > > may be happening and how I can prevent it? > > > > Many thanks > > Andy > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/7ff465e6/attachment.html From solko at gcdf.pl Sat Mar 21 01:29:38 2009 From: solko at gcdf.pl (Szymon Olko) Date: Sat, 21 Mar 2009 09:29:38 +0100 Subject: [Freeswitch-users] proper way to clean source tree after build? In-Reply-To: <20090321021028.GA5399@jdc.jasonjgw.net> References: <20090321021028.GA5399@jdc.jasonjgw.net> Message-ID: <49C4A572.9070803@gcdf.pl> Jason White pisze: > I'm building under Debian. > > To clean the source tree after building, I've tried > make clean > debuild clean > > but what often happens is that after updating to a later svn revision and then > trying to build, I get various errors during the compilation process that > disappear if I make a clean checkout from the repository and run the build > again. > > I have also found that using svn export to reproduce the tree in another > directory, then building from that directory, doesn't always solve the problem. > > Is there a better way to clean up the source tree after updating it, to avoid > these issues? I assume there are extraneous modifications or files that simply > aren't being eliminated by the "clean" makefile targets. > > There is 'make current' which make cleaning, uninstalling, updating, building and installing new version. I looked into make file, and I'm doing all without uninstall and install, which I call later when it is possible for me. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Sun Mar 22 23:31:00 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 23 Mar 2009 17:31:00 +1100 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles Message-ID: <20090323063100.GA5058@jdc.jasonjgw.net> I have TLS enabled in my internal and internal-ipv6 profiles as per the stock configuration. When FreeSWITCH is started, sometimes either of the profiles fails to initialize, with an "Unable to create SIP UA for profile" error in the log. If I then start the profile manually sofia profile start from fs_cli, the profile starts up as it should. So far, this has only occurred with the profiles for which I have TLS enabled. I can do more testing to see whether that's part of the problem. Meanwhile, has anyone else seen this? It's revision 12701 from svn. From zhaoxxqq at 163.com Mon Mar 23 02:17:20 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Mon, 23 Mar 2009 17:17:20 +0800 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 Message-ID: <200903231717175620353@163.com> HI, friend, I added to my sip profile in external , like below. --> but. the problem is still exist. Can you help me. Zhao Xiaoqiang ------------------------------------------------------------------------------------------------------------ Hi - It looks like you're getting digits both in the RTP stream and as SIP INFO. Try adding to the SIP profile you're using for inbound calls. --Dave > I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS > to PSTN with DID numbers. For inband I connect it to FS's demo_ivr. > When I press the key, the FS accept always DOUBLE of key. The debug > information like below. > > 2009-03-22 17:50:26 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 > 2009-03-22 17:50:26 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file > 2009-03-22 17:50:26 [DEBUG] switch_ivr_menu.c:308 play_and_collect() waiting for 3/4 digits t/o 2000 > 2009-03-22 17:50:26 [DEBUG] sofia.c:3753 sofia_handle_sip_i_info() INFO DTMF(1) > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:353 play_and_collect() digits '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:523 switch_ivr_menu_execute() IVR menu 'jtq_greating' caught invalid input '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-03-22 17:50:28 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/external/13323015 > > > Can any friend can help me? > > Zhao Xiaoqiang > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/af706274/attachment.html From Claudio.Cavalera at italtel.it Mon Mar 23 02:17:53 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Mon, 23 Mar 2009 10:17:53 +0100 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <22628039.post@talk.nabble.com> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > Yes that page was perfect, ty. > > (Cepstral 5.1) As far as I know the use of Cepstral 5.x is discouraged and it should be better to stick with Cepstral 4. Unfortunately Cepstral 4 voices are hard to find. :-\ BRs, Claudio Internet E. Mail Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. From brian at freeswitch.org Mon Mar 23 04:53:49 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 06:53:49 -0500 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles In-Reply-To: <20090323063100.GA5058@jdc.jasonjgw.net> References: <20090323063100.GA5058@jdc.jasonjgw.net> Message-ID: You didn't include enough info ... Distro, OS or sofia log... It could be many things... socket lingering is my best guess. /b On Mar 23, 2009, at 1:31 AM, Jason White wrote: > I have TLS enabled in my internal and internal-ipv6 profiles as per > the stock > configuration. > > When FreeSWITCH is started, sometimes either of the profiles fails to > initialize, with an "Unable to create SIP UA for profile" error in > the log. If > I then start the profile manually > sofia profile start > from fs_cli, the profile starts up as it should. > > So far, this has only occurred with the profiles for which I have > TLS enabled. > I can do more testing to see whether that's part of the problem. > > Meanwhile, has anyone else seen this? > > It's revision 12701 from svn. From brian at freeswitch.org Mon Mar 23 04:54:34 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 06:54:34 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: <200903231717175620353@163.com> References: <200903231717175620353@163.com> Message-ID: <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> Tell your phone to stop sending INFO and 2833 at the same time and the problem will stop. /b On Mar 23, 2009, at 4:17 AM, zhaoxxqq wrote: > HI, friend, > I added to my sip profile > in external , like below. > > --> > > > > > > > > but. the problem is still exist. Can you help me. > > Zhao Xiaoqiang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/17036583/attachment.html From dave at 3c.co.uk Mon Mar 23 05:47:16 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 23 Mar 2009 06:47:16 -0600 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> References: <200903231717175620353@163.com> <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> Message-ID: <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> Sorry - my bad - dtmf-type looks like it just controls what's sent, not what's received. Brian's advice is sound, or you can probably work around things right now by editing src/mod/endpoints/mod_sofia/sofia.c - at around line 3838 you'll find: if (dtmf.digit) { /* queue it up */ switch_channel_queue_dtmf(channel, &dtmf); ..more code.. /* Send 200 OK response */ nua_respond(nh, SIP_200_OK, NUTAG_WITH_THIS(nua), TAG_END()); - lose the bit which handles the SIP INFO DTMF by adding a couple of lines thusly: if (dtmf.digit) { #if 0 /* queue it up */ switch_channel_queue_dtmf(channel, &dtmf); ..more code.. #endif /* Send 200 OK response */ nua_respond(nh, SIP_200_OK, NUTAG_WITH_THIS(nua), TAG_END()); It's a nasty hack, but it just might work. --Dave > Tell your phone to stop sending INFO and 2833 at the same time and > the problem will stop. > > /b > > On Mar 23, 2009, at 4:17 AM, zhaoxxqq wrote: > >> HI, friend, >> I added to my sip >> profile in external , like below. >> >> --> >> >> >> >> >> >> >> >> but. the problem is still exist. Can you help me. >> >> Zhao Xiaoqiang > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/851640ad/attachment.html From brian at freeswitch.org Mon Mar 23 06:10:34 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 08:10:34 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> References: <200903231717175620353@163.com> <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> Message-ID: It's more sane to have the phone to NOT send them both in the first place because it is WRONG to send both info and 2833 and NOT totally expect the far end to make heads or tails of it. How about actually have the phone manufacture fix their broken phone? /b On Mar 23, 2009, at 7:47 AM, David Knell wrote: > > It's a nasty hack, but it just might work. > > --Dave From gerry at pstn2.net Mon Mar 23 07:36:41 2009 From: gerry at pstn2.net (Gerry Hull) Date: Mon, 23 Mar 2009 10:36:41 -0400 Subject: [Freeswitch-users] Two portaudio UA's, one FS. Possible? Message-ID: <98a86adf0903230736m66a32ee7u44db2ded80d15c81@mail.gmail.com> I have an app where I would like to run two portaudio user agents on the same computer (two sound cards). I want one UA to be feeding a conference, and the other as a softphone. I don't see a way to run two portaudio ua's on the same instance of FS. Is this possible. If not, OK to run two FS instances? TIA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/a8681a0d/attachment.html From anthony.minessale at gmail.com Mon Mar 23 08:14:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Mar 2009 10:14:44 -0500 Subject: [Freeswitch-users] Two portaudio UA's, one FS. Possible? In-Reply-To: <98a86adf0903230736m66a32ee7u44db2ded80d15c81@mail.gmail.com> References: <98a86adf0903230736m66a32ee7u44db2ded80d15c81@mail.gmail.com> Message-ID: <191c3a030903230814g14533908u5ccc2211a6453303@mail.gmail.com> it's not supported to run 2 instances of PA on 2 soundcards at once in one FS, but it is ok to run 2 difft copies of FS if each one has its own config log and db dirs and none of the configuration collides. 2009/3/23 Gerry Hull > I have an app where I would like to run two portaudio user agents on the > same computer (two sound cards). I want one UA to be feeding a conference, > and the other as a softphone. I don't see a way to run two portaudio ua's > on the same instance of FS. Is this possible. If not, OK to run two FS > instances? > > TIA > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/cc111a45/attachment.html From dave at 3c.co.uk Mon Mar 23 08:33:06 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 23 Mar 2009 09:33:06 -0600 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: References: <200903231717175620353@163.com> <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> Message-ID: <49C7ABB2.80606@3c.co.uk> Hi Brian, > It's more sane to have the phone to NOT send them both in the first > place because it is WRONG to send both info and 2833 and NOT totally > expect the far end to make heads or tails of it. > > How about actually have the phone manufacture fix their broken phone? > In an ideal world, of course; however:- (a) the quick hack is probably a path of lesser resistance to getting Zhao up and running with FS; (b) he said it was an inbound SIP provider, rather than a phone, that he was using, so he'd need to get them to fix their end: might be trivial, might not. --Dave From reply at matthewfong.com Mon Mar 23 08:49:40 2009 From: reply at matthewfong.com (Matthew Fong) Date: Mon, 23 Mar 2009 22:49:40 +0700 Subject: [Freeswitch-users] Another fifo request Message-ID: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/e6462470/attachment.html From mattdfong at gmail.com Mon Mar 23 08:50:16 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Mon, 23 Mar 2009 22:50:16 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> Message-ID: <4256bf830903230850p100b44ffm52e6292fd281aa5@mail.gmail.com> Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/44816f63/attachment.html From anthony.minessale at gmail.com Mon Mar 23 08:53:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Mar 2009 10:53:31 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: <49C7ABB2.80606@3c.co.uk> References: <200903231717175620353@163.com> <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> <49C7ABB2.80606@3c.co.uk> Message-ID: <191c3a030903230853i69bd114dg2fa64ee04ca2976b@mail.gmail.com> c) if we have to, we could add a patch to pick which types of dtmf to accept as well so he could force the equiv of the hack. On Mon, Mar 23, 2009 at 10:33 AM, David Knell wrote: > Hi Brian, > > It's more sane to have the phone to NOT send them both in the first > > place because it is WRONG to send both info and 2833 and NOT totally > > expect the far end to make heads or tails of it. > > > > How about actually have the phone manufacture fix their broken phone? > > > In an ideal world, of course; however:- > (a) the quick hack is probably a path of lesser resistance to getting > Zhao up > and running with FS; > (b) he said it was an inbound SIP provider, rather than a phone, that he > was > using, so he'd need to get them to fix their end: might be trivial, > might not. > > --Dave > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/38d727df/attachment.html From anthony.minessale at gmail.com Mon Mar 23 09:08:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Mar 2009 11:08:05 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> Message-ID: <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> ok, maybe after this i can have a day off ;) 2 variables added to latest trunk: "fifo_caller_consumer_import" "fifo_consumer_caller_import" both work like the regular import but one is a list of vars to copy from caller to consumer and one is a list to copy from consumer to caller. 2009/3/23 Matthew Fong > Thanks Anthony, for creating the transfer_after_bridge feature for me. > Your rapid actions, are the reason I'm positive I made the right decision > switch to to FS. > I got another challenge to throw your way. In the current fifo > implementation, there's no way to identify which fifo consumer, consumes a > caller--besides using other_leg_unique_id on bridge (ie, there's no way to > pass data between channels when a fifo bridge is created). I want to be able > to transfer some caller information to the consumer channel on bridge, to > populate an agent's screen. > > Under normal (non-fifo) circumstances, when a call is bridged, I've used > the 'import' channel variable, so that onBridge, the aleg automatically gets > populated with the bleg's 'import' field. However when fifo bridges, it does > not recognize import. In other applications, I've gotten around this by > using bridge_pre_execute_bleg_app/data to throw a custom event but with > fifo, bridge_pre_execute also does not work. I've been looking at the > fifo::info event, but again, there's no fifo_action that directly links > caller variables and consumer variables together. > > For now at least, I can get around this by storing uuid information in my > separate database, and looking up the other channel's information based > on other_leg_unique_id, but it would be nice if I could directly see it when > the channel is bridged. Anyway, great program, and I hope you consider > implementing these features to make FS even better. Thanks. > > --matt > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/73636456/attachment-0001.html From qulix at mail.ru Mon Mar 23 09:31:59 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Mon, 23 Mar 2009 19:31:59 +0300 Subject: [Freeswitch-users] Inbound calls Message-ID: Hi all! There is an error if I try to make inbound call. I don't think it's hard but I'm tired solving it. Any time I call my city number, FS always returns me an error [INCOMPATIBLE_DESTINATION] 2009-03-23 00:29:40 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/mydomain/number-ani at ip [8d0dcff8-1728-11de-b150-e189b327b8ae] 2009-03-23 00:29:40 [NOTICE] sofia.c:2927 sofia_handle_sip_i_state() Hangup sofia/mydomain/number-ani at ip [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-03-23 00:29:40 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 2 (sofia/mydomain/number-ani at ip) Ended 2009-03-23 00:29:40 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/mydomain/number-ani at ip [CS_HANGUP] My extension is: I've made it all the same as http://wiki.freeswitch.org/wiki/Home_PBX_Example (with my own diference line inbound number, dest_number , etc) Could you share some expirience? Give any solution? Or link where I could learn more about it? =\ From brian at freeswitch.org Mon Mar 23 09:39:50 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 11:39:50 -0500 Subject: [Freeswitch-users] Inbound calls In-Reply-To: References: Message-ID: <46DE35FD-1DFC-45F1-9BA6-106EA65022AD@freeswitch.org> make current I think you might have snagged the rev over the weekend while we were fixing a bug. /b On Mar 23, 2009, at 11:31 AM, ????... wrote: > Any time I call my city number, FS always returns me an error > [INCOMPATIBLE_DESTINATION] From codecomplete at free.fr Mon Mar 23 11:45:17 2009 From: codecomplete at free.fr (Gilles) Date: Mon, 23 Mar 2009 19:45:17 +0100 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error Message-ID: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> Hello, This is my first try at compiling Freeswitch (1.0.3). Since I have a Linux box lying around, I'm giving it a shot on a "SUSE Linux Enterprise Desktop 10 SP1 (i586)". Per the instructions in the wiki, I run "make", but get the following error : http://pastebin.ca/1369381 According to the archives, it's likely due to the missing ODBC-devel package. I have a couple of questions: 1. Does Freeswitch require ODBC? 2. Please forgive the newbie question, but... how do I find and install this package on Suse? Thank you. From grevenx at me.com Mon Mar 23 12:01:56 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Mon, 23 Mar 2009 20:01:56 +0100 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error In-Reply-To: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> References: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> Message-ID: 1. FS does not require ODBC, uses SQLite per default How did you run your ./configure ? My guess is that you either have choosen to compile with ODBC, or that the build system somehow detects that you have ODBC installed and tries to compile it with ODBC enabled (and fails). Best regard, Even Andr? Fiskvik On 23. mars. 2009, at 19.45, Gilles wrote: > Hello, > > This is my first try at compiling Freeswitch (1.0.3). Since I have a > Linux box lying around, I'm giving it a shot on a "SUSE Linux > Enterprise Desktop 10 SP1 (i586)". > > Per the instructions in the wiki, I run "make", but get the following > error : http://pastebin.ca/1369381 > > According to the archives, it's likely due to the missing ODBC-devel > package. I have a couple of questions: > 1. Does Freeswitch require ODBC? > 2. Please forgive the newbie question, but... how do I find and > install this package on Suse? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Mar 23 12:22:36 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 14:22:36 -0500 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error In-Reply-To: References: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> Message-ID: <23CC9072-8B9E-4BCB-9B12-91F2FF22BCBE@freeswitch.org> We changed this to autodetect it means you have the libs but no devel headers.. either disable it or install the dev headers. /b On Mar 23, 2009, at 2:01 PM, Even Andr? Fiskvik wrote: > 1. FS does not require ODBC, uses SQLite per default > > How did you run your ./configure ? > My guess is that you either have choosen to compile with ODBC, or that > the > build system somehow detects that you have ODBC installed and tries to > compile it > with ODBC enabled (and fails). > > Best regard, > Even Andr? Fiskvik From raul at etellicom.com Mon Mar 23 12:31:44 2009 From: raul at etellicom.com (Raul Fragoso) Date: Mon, 23 Mar 2009 16:31:44 -0300 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error In-Reply-To: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> References: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> Message-ID: <1237836704.28992.1.camel@raul-laptop> Running the following command as root should install the ODBC development package: yast install unixODBC-devel After that, run configure again and make FS as usual. Regards, Raul On Mon, 2009-03-23 at 19:45 +0100, Gilles wrote: > Hello, > > This is my first try at compiling Freeswitch (1.0.3). Since I have a > Linux box lying around, I'm giving it a shot on a "SUSE Linux > Enterprise Desktop 10 SP1 (i586)". > > Per the instructions in the wiki, I run "make", but get the following > error : http://pastebin.ca/1369381 > > According to the archives, it's likely due to the missing ODBC-devel > package. I have a couple of questions: > 1. Does Freeswitch require ODBC? > 2. Please forgive the newbie question, but... how do I find and > install this package on Suse? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Mar 23 12:43:43 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Mar 2009 15:43:43 -0400 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error In-Reply-To: <1237836704.28992.1.camel@raul-laptop> References: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> <1237836704.28992.1.camel@raul-laptop> Message-ID: <956825C4-39EF-4E12-A96C-0C6B37DB19EE@jerris.com> I still need access to a machine in this state so I can debug and fix this issue. Mike On Mar 23, 2009, at 3:31 PM, Raul Fragoso wrote: > Running the following command as root should install the ODBC > development package: > yast install unixODBC-devel > > After that, run configure again and make FS as usual. > > Regards, > > Raul > > On Mon, 2009-03-23 at 19:45 +0100, Gilles wrote: >> Hello, >> >> This is my first try at compiling Freeswitch (1.0.3). Since I have a >> Linux box lying around, I'm giving it a shot on a "SUSE Linux >> Enterprise Desktop 10 SP1 (i586)". >> >> Per the instructions in the wiki, I run "make", but get the following >> error : http://pastebin.ca/1369381 >> >> According to the archives, it's likely due to the missing ODBC-devel >> package. I have a couple of questions: >> 1. Does Freeswitch require ODBC? >> 2. Please forgive the newbie question, but... how do I find and >> install this package on Suse? >> >> Thank you. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch-users at digitaldan.com Mon Mar 23 14:14:45 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Mon, 23 Mar 2009 15:14:45 -0600 (MDT) Subject: [Freeswitch-users] Sip for Skype Message-ID: <13257395.221237842869198.JavaMail.daniel@radio> You probably already saw this but.... http://www.skypeforsip.com/ Skype is supporting sip for business users. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/bdc3369a/attachment.html From pabx_freeswitch at telenet.be Mon Mar 23 15:22:42 2009 From: pabx_freeswitch at telenet.be (henkoegema) Date: Mon, 23 Mar 2009 15:22:42 -0700 (PDT) Subject: [Freeswitch-users] Different files (?) Message-ID: <22670515.post@talk.nabble.com> What is the difference between between the following 3 files: 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 26M 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M Which one should I download, if I want the latest (newest) ? -- View this message in context: http://www.nabble.com/Different-files-%28-%29-tp22670515p22670515.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From pabx_freeswitch at telenet.be Mon Mar 23 15:24:23 2009 From: pabx_freeswitch at telenet.be (henkoegema) Date: Mon, 23 Mar 2009 15:24:23 -0700 (PDT) Subject: [Freeswitch-users] Different files (?) Message-ID: <22670515.post@talk.nabble.com> What is the difference between the following 3 files: 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 26M 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M Which one should I download, if I want the latest (newest) ? -- View this message in context: http://www.nabble.com/Different-files-%28-%29-tp22670515p22670515.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Mar 23 15:34:43 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 17:34:43 -0500 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <22670515.post@talk.nabble.com> References: <22670515.post@talk.nabble.com> Message-ID: <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> I would highly recommend you use the snapshot or svn checkout because we are close to 1.0.4 and it looks like the symlink wasn't updated for 1.0.3 we'll get that fixed. /b On Mar 23, 2009, at 5:24 PM, henkoegema wrote: > 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 > 26M > 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M > 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M From pablosaro at gmail.com Mon Mar 23 16:19:43 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Mon, 23 Mar 2009 20:19:43 -0300 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> References: <22670515.post@talk.nabble.com> <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> Message-ID: <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> BTW, you also recommend SVN trunk for production servers? IMHO, should be a stable release for this purpose. On 3/23/09, Brian West wrote: > I would highly recommend you use the snapshot or svn checkout because > we are close to 1.0.4 and it looks like the symlink wasn't updated for > 1.0.3 we'll get that fixed. > > /b > > On Mar 23, 2009, at 5:24 PM, henkoegema wrote: > >> 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 >> 26M >> 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M >> 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from Gmail for mobile | mobile.google.com From msc at freeswitch.org Mon Mar 23 16:24:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Mar 2009 16:24:49 -0700 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> References: <22670515.post@talk.nabble.com> <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> Message-ID: <87f2f3b90903231624q563b6f1dw1075f30b414f27fe@mail.gmail.com> On Mon, Mar 23, 2009 at 4:19 PM, Pablo Hernan Saro wrote: > BTW, you also recommend SVN trunk for production servers? > IMHO, should be a stable release for this purpose. FreeSWITCH is one of those unusual projects where the SVN trunk is generally more stable than the tagged releases. -MC From brian at freeswitch.org Mon Mar 23 16:26:41 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 18:26:41 -0500 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> References: <22670515.post@talk.nabble.com> <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> Message-ID: <0C8A5F39-868C-478B-931C-B316705EB939@freeswitch.org> Thats up to you... I use SVN Trunk when I know things are good but thats my job... thats why I work hard on QA for the project. We did have some regressions over the past few days that we did finally narrow down and fix over the weekend... and yes we work pretty much 24/7 :) We are trying to get SVN to the point so we can release 1.0.4 and it has MANY fixes over 1.0.3. /b On Mar 23, 2009, at 6:19 PM, Pablo Hernan Saro wrote: > BTW, you also recommend SVN trunk for production servers? > IMHO, should be a stable release for this purpose. > From krice at suspicious.org Mon Mar 23 16:31:07 2009 From: krice at suspicious.org (Ken Rice) Date: Mon, 23 Mar 2009 18:31:07 -0500 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> Message-ID: I run SVN on several production servers... "Stable Releases" are just that some point in the code where the maintainers felt it was stable enuff to release on that given day... With FreeSwitch however it never fails, a big bug is found and fixed w/in 2 days of that "stable release" so its not really that stable at all... Until we get more people testing and actually reporting bugs on RCs I doubt this well ever change... > From: Pablo Hernan Saro > Reply-To: > Date: Mon, 23 Mar 2009 20:19:43 -0300 > To: > Subject: Re: [Freeswitch-users] Different files (?) > > BTW, you also recommend SVN trunk for production servers? > IMHO, should be a stable release for this purpose. > > > > On 3/23/09, Brian West wrote: >> I would highly recommend you use the snapshot or svn checkout because >> we are close to 1.0.4 and it looks like the symlink wasn't updated for >> 1.0.3 we'll get that fixed. >> >> /b >> >> On Mar 23, 2009, at 5:24 PM, henkoegema wrote: >> >>> 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 >>> 26M >>> 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M >>> 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- > Sent from Gmail for mobile | mobile.google.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Mar 23 16:32:58 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 18:32:58 -0500 Subject: [Freeswitch-users] Different files (?) In-Reply-To: References: Message-ID: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> Well put! /b On Mar 23, 2009, at 6:31 PM, Ken Rice wrote: > > Until we get more people testing and actually reporting bugs on RCs > I doubt > this well ever change... From jason at jasonjgw.net Mon Mar 23 16:56:46 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 24 Mar 2009 10:56:46 +1100 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles In-Reply-To: <20090323063100.GA5058@jdc.jasonjgw.net> References: <20090323063100.GA5058@jdc.jasonjgw.net> Message-ID: <20090323235646.GA11567@jdc.jasonjgw.net> Here's a log. OS: Debian Sid, kernel 2.6.26. This happens when FreeSWITCH is started during the boot process from the init scripts, as well as subsequently if it is shut down and restarted. Sometimes it is the internal profile that fails, rather than internal-ipv6, as here. tport_tls_init_master(0x18f6bd0): tls key = /opt/freeswitch/conf/ssl/agent.pem tls_init_context: 140a90a1:SSL routines:(null):(null) tport_listen(0x7fedc0006960): tls_init_master(pf=10 tls/[2001:470:801f::2]:5061): Input/output error nta: bind([2001:470:801f::2]:5061;transport=tls): Input/output error nua: initializing SIP stack failed nua: nua_stack_deinit: entering tport_destroy(0x7fedc0006960) su_epoll_port_deinit(0x18fa810) called 2009-03-24 10:42:12 [ERR] sofia.c:750 sofia_profile_thread_run() Error Creating SIP UA for profile: internal-ipv6 It looks like an operating system issue to me. From mszlazak at aol.com Mon Mar 23 23:40:57 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 24 Mar 2009 02:40:57 -0400 Subject: [Freeswitch-users] setInputCallback not working with Javascript? Message-ID: <8CB7A629F5159D9-17A0-AA5@webmail-de13.sysops.aol.com> I'm getting in build 12653M: [ERR] notify.js:130 mod_spidermonkey()? TypeError: session.setInputCallback is not a function The wiki says this function should work in Javascript. http://wiki.freeswitch.org/wiki/CoreSession_Constructor#session:setInputCallback Also, has there been changes to session.collectInput with type="event"? I get dtmf type events with my callback function but can't seem to get type="event" with speech events. Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/14f2d606/attachment.html From codecomplete at free.fr Tue Mar 24 02:15:09 2009 From: codecomplete at free.fr (Gilles) Date: Tue, 24 Mar 2009 10:15:09 +0100 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error Message-ID: <7.0.1.0.2.20090324095733.0242cf88@fredshack.com> Brian West > We changed this to autodetect it means you have the libs but no devel headers.. either disable it or install the dev headers. Thanks guys, that was it: The stock SLED 10 that I have on this MSI host did have unixODBC.rpm but not its unixODBC-devel.rpm counterpart. I re-read the "Download & Installation Guide" page, which doesn't say that Freeswitch will compile for ODBC instead of SQLite if it detects unixODBC... but doesn't check that its counterpart unixODBC-devel is also installed :-/ I suggest that either the config/make script be updated to check for this, or the wiki be edited to tell users to check for both packages (if N.A. : yast install unixODBC-devel OR rpm -UVH unixODBC-devel.rpm) or configure Freeswitch to use SQLite instead of ODBC (good enough to get newbies started). Cheers, From willbelair at yahoo.com Tue Mar 24 05:51:57 2009 From: willbelair at yahoo.com (Will Smith) Date: Tue, 24 Mar 2009 05:51:57 -0700 (PDT) Subject: [Freeswitch-users] Console Window Message-ID: <996845.60281.qm@web53607.mail.re2.yahoo.com> Hi, I closed the console window that ran FS without shutting down FS. I had to restart the server to get back to the console window. I just wonder if there is a way to bring that window up without restart server. In the file /log/freeswitch.pid, I found a number, is that a seesion id? ? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/a87e6136/attachment.html From dyfet at gnutelephony.org Tue Mar 24 05:54:07 2009 From: dyfet at gnutelephony.org (David Sugar) Date: Tue, 24 Mar 2009 08:54:07 -0400 Subject: [Freeswitch-users] Sip for Skype - g.729 requirement In-Reply-To: <13257395.221237842869198.JavaMail.daniel@radio> References: <13257395.221237842869198.JavaMail.daniel@radio> Message-ID: <49C8D7EF.8050106@gnutelephony.org> They require one use g.729, which is patent encumbered as well as rather computationally intensive. Dan wrote: > You probably already saw this but.... > > http://www.skypeforsip.com/ > > Skype is supporting sip for business users. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: dyfet.vcf Type: text/x-vcard Size: 186 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/6452a43a/attachment.vcf From saeedahmad1981 at gmail.com Tue Mar 24 06:25:08 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Tue, 24 Mar 2009 14:25:08 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <996845.60281.qm@web53607.mail.re2.yahoo.com> References: <996845.60281.qm@web53607.mail.re2.yahoo.com> Message-ID: <19BE75BD4A6543468D8EA83AF14580CC@SaeedLaptop> Did you started FS with -nc option? with this option you can connect to FS using ./fs_cli OR use screen! _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Will Smith Sent: Tuesday, March 24, 2009 1:52 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Console Window Hi, I closed the console window that ran FS without shutting down FS. I had to restart the server to get back to the console window. I just wonder if there is a way to bring that window up without restart server. In the file /log/freeswitch.pid, I found a number, is that a seesion id? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/0ca2c99f/attachment-0001.html From willbelair at yahoo.com Tue Mar 24 06:35:09 2009 From: willbelair at yahoo.com (Will Smith) Date: Tue, 24 Mar 2009 06:35:09 -0700 (PDT) Subject: [Freeswitch-users] Console Window In-Reply-To: <19BE75BD4A6543468D8EA83AF14580CC@SaeedLaptop> Message-ID: <308674.85493.qm@web53611.mail.re2.yahoo.com> No, I started FS with this: /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? ? thank you for help --- On Tue, 3/24/09, Saeed Ahmed wrote: From: Saeed Ahmed Subject: Re: [Freeswitch-users] Console Window To: freeswitch-users at lists.freeswitch.org Date: Tuesday, March 24, 2009, 6:25 AM Did you started FS with ?nc option? with this option you can connect to FS using ./fs_cli OR use screen! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Will Smith Sent: Tuesday, March 24, 2009 1:52 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Console Window ? Hi, I closed the console window that ran FS without shutting down FS. I had to restart the server to get back to the console window. I just wonder if there is a way to bring that window up without restart server. In the file /log/freeswitch.pid, I found a number, is that a seesion id? ? Thank you. ?_______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/1a0bb79f/attachment.html From steve.d.ward at gmail.com Tue Mar 24 06:42:54 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 09:42:54 -0400 Subject: [Freeswitch-users] sip cancel request fails Message-ID: <4ea6e8f20903240642s1c8f46afld96291ca8fb51377@mail.gmail.com> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while the call is still ringing does not work. Why is this request resulting in a 481? I appreciate the help - I'm still just starting to learn SIP & FS. The CANCEL request and 481 response appear as follows on my FS console: recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: ------------------------------------------------------------------------ CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport From: "Steve" >;tag=as7f6965ea To: > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 From: "Steve" >;tag=as7f6965ea To: >;tag=71m745HKHKyjc Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 CSeq: 103 CANCEL Content-Length: 0 -------------------------------------- Thanks. - SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/82662ad8/attachment.html From solko at gcdf.pl Tue Mar 24 06:42:56 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 24 Mar 2009 14:42:56 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <308674.85493.qm@web53611.mail.re2.yahoo.com> References: <308674.85493.qm@web53611.mail.re2.yahoo.com> Message-ID: <49C8E360.8050709@gcdf.pl> Will Smith pisze: > No, I started FS with this: > /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? > run command fs_cli it is almost the same as normal freeswitch console. You just cannot shut it down with '...' command. Just use this one 'fsctl shutdown asap'. > thank you for help > > --- On *Tue, 3/24/09, Saeed Ahmed //* wrote: > > From: Saeed Ahmed > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:25 AM > > Did you started FS with ?nc option? > with this option you can connect to FS using ./fs_cli > OR > use /screen/! > > ------------------------------------------------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Will Smith > *Sent:* Tuesday, March 24, 2009 1:52 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Console Window > > > > Hi, > > I closed the console window that ran FS without shutting down FS. I > had to restart the server to get back to the console window. I just > wonder if there is a way to bring that window up without restart > server. In the file /log/freeswitch.pid, I found a number, is that a > seesion id? > > > > Thank you. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steve.d.ward at gmail.com Tue Mar 24 06:43:12 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 09:43:12 -0400 Subject: [Freeswitch-users] sip cancel request fails Message-ID: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while the call is still ringing does not work. Why is this request resulting in a 481? I appreciate the help - I'm still just starting to learn SIP & FS. The CANCEL request and 481 response appear as follows on my FS console: recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: ------------------------------------------------------------------------ CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport From: "Steve" >;tag=as7f6965ea To: > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 From: "Steve" >;tag=as7f6965ea To: >;tag=71m745HKHKyjc Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 CSeq: 103 CANCEL Content-Length: 0 -------------------------------------- Thanks. - SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/fe31b242/attachment.html From brian at freeswitch.org Tue Mar 24 06:47:01 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Mar 2009 08:47:01 -0500 Subject: [Freeswitch-users] setInputCallback not working with Javascript? In-Reply-To: <8CB7A629F5159D9-17A0-AA5@webmail-de13.sysops.aol.com> References: <8CB7A629F5159D9-17A0-AA5@webmail-de13.sysops.aol.com> Message-ID: <554DE828-B96A-4E17-A974-151CAC50A0E5@freeswitch.org> Javascript doesn't use the Core Session constructor. Its not the same as the other languages. /b On Mar 24, 2009, at 1:40 AM, mszlazak at aol.com wrote: > I'm getting in build 12653M: > > [ERR] notify.js:130 mod_spidermonkey() TypeError: > session.setInputCallback is not a function > > The wiki says this function should work in Javascript. > > http://wiki.freeswitch.org/wiki/ > CoreSession_Constructor#session:setInputCallback > > Also, has there been changes to session.collectInput with > type="event"? I get dtmf type events with my callback function but > can't seem to get type="event" with speech events. > > Mark. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/ad67a971/attachment.html From leon at scarlet-internet.nl Tue Mar 24 06:47:15 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 24 Mar 2009 14:47:15 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <308674.85493.qm@web53611.mail.re2.yahoo.com> References: <308674.85493.qm@web53611.mail.re2.yahoo.com> Message-ID: Just make sure you have mod_event socket loaded in conf/ autoload_modules/modules.conf.xml : And have it configured in conf/autoload_modules/event_socket.conf.xml Then you can use bin/fs_cli to connect to running FS instance. I don't think you can reconnect to a process of which you have disconnected the terminal (without using screen).. regards, Leon On Mar 24, 2009, at 2:35 PM, Will Smith wrote: > No, I started FS with this: > /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? > > thank you for help > > --- On Tue, 3/24/09, Saeed Ahmed wrote: > From: Saeed Ahmed > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:25 AM > > Did you started FS with ?nc option? > with this option you can connect to FS using ./fs_cli > OR > use screen! > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Will Smith > Sent: Tuesday, March 24, 2009 1:52 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Console Window > > > Hi, > > I closed the console window that ran FS without shutting down FS. I > had to restart the server to get back to the console window. I just > wonder if there is a way to bring that window up without restart > server. In the file /log/freeswitch.pid, I found a number, is that a > seesion id? > > > > Thank you. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/eadef7e7/attachment-0001.html From mike at jerris.com Tue Mar 24 06:47:54 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Mar 2009 09:47:54 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> Message-ID: This means we could not match the cancel to a current call dialog. I would need to see the full sip trace of the call to know why, but typically this is because of not matching call Id or to or from tags Mike On Mar 24, 2009, at 9:43 AM, Steven Ward wrote: > A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b- > lab-1) while the call is still ringing does not work. > > Why is this request resulting in a 481? > > I appreciate the help - I'm still just starting to learn SIP & FS. > The CANCEL request and 481 response appear as follows on my FS > console: > > > recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: > > --- > --------------------------------------------------------------------- > CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport > From: "Steve" ;tag=as7f6965ea > To: > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > --- > --------------------------------------------------------------------- > send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: > > --- > --------------------------------------------------------------------- > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 > From: "Steve" ;tag=as7f6965ea > To: ;tag=71m745HKHKyjc > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 > CSeq: 103 CANCEL > Content-Length: 0 > > -------------------------------------- > > > > Thanks. - SW > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/63049bf8/attachment.html From steve.d.ward at gmail.com Tue Mar 24 06:57:10 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 09:57:10 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> Message-ID: <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> Here it is: freeswitch at b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 at 13:53:07.644865: ------------------------------------------------------------------------ OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport From: "Unknown" >;tag=as5adee8f4 To: Contact: > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 24 Mar 2009 13:53:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ------------------------------------------------------------------------ send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060 From: "Unknown" >;tag=as5adee8f4 To: ;tag=DytraHp3K84aD Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 CSeq: 102 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169: ------------------------------------------------------------------------ INVITE sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport From: "Steve" >;tag=as4863e49a To: > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 24 Mar 2009 13:53:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 258 v=0 o=root 4756 4756 IN IP4 10.1.21.44 s=session c=IN IP4 10.1.21.44 t=0 0 m=audio 17956 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 From: "Steve" >;tag=as4863e49a To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Content-Length: 0 ------------------------------------------------------------------------ send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 From: "Steve" >;tag=as4863e49a To: >;tag=e7KHcc76gHUXr Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="10.1.21.44", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103: ------------------------------------------------------------------------ ACK sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport From: "Steve" >;tag=as4863e49a To: >;tag=e7KHcc76gHUXr Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306: ------------------------------------------------------------------------ INVITE sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport From: "Steve" >;tag=as4863e49a To: > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="b-pbx-lab-1", realm="10.1.21.44", algorithm=MD5, uri="sip:70904 at b-pbx-lab-1.mynet.net", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, cnonce="0e89cc90", nc=00000001 Date: Tue, 24 Mar 2009 13:53:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 258 v=0 o=root 4756 4757 IN IP4 10.1.21.44 s=session c=IN IP4 10.1.21.44 t=0 0 m=audio 17956 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 From: "Steve" >;tag=as4863e49a To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Content-Length: 0 ------------------------------------------------------------------------ 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/70904 at 10.1.21.44[1d28557e-187b-11de-8c60-ad87768304bc] 2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing Steve->70904 in context default 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes [1d3a376c-187b-11de-8c60-ad87768304bc] send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291: ------------------------------------------------------------------------ INVITE sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c SIP/2.0 Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p Max-Forwards: 69 From: "Steve" >;tag=gS62F28DB372F To: Call-ID: f4992499-931d-122c-34b1-003018ae1862 CSeq: 112833059 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 328 Remote-Party-ID: "Steve" >;screen=yes;privacy=off v=0 o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 10.1.21.45 s=FreeSWITCH c=IN IP4 10.1.21.45 t=0 0 m=audio 22432 RTP/AVP 0 9 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p Contact: To: ;tag=fa138551 From: "Steve" >;tag=gS62F28DB372F Call-ID: f4992499-931d-122c-34b1-003018ae1862 CSeq: 112833059 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ------------------------------------------------------------------------ 2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:70904 at 10.1.56.106:44952 ;rinstance=481ff1bdc7ab2a4c;fs_nat=yes! send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 From: "Steve" >;tag=as4863e49a To: >;tag=FgDae7QaetHgm Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 sofia_receive_message() Ring-Ready sofia/internal/70904 at 10.1.21.44! 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/70904 at 10.1.21.44! recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013: ------------------------------------------------------------------------ CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport From: "Steve" >;tag=as4863e49a To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060 From: "Steve" >;tag=as4863e49a To: >;tag=FgDae7QaetHgm Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 CANCEL Content-Length: 0 ------------------------------------------------------------------------ 2009/3/24 Michael Jerris > This means we could not match the cancel to a current call dialog. I > would need to see the full sip trace of the call to know why, but typically > this is because of not matching call Id or to or from tags > > Mike > > > On Mar 24, 2009, at 9:43 AM, Steven Ward wrote: > > A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) > while the call is still ringing does not work. > > Why is this request resulting in a 481? > > I appreciate the help - I'm still just starting to learn SIP & FS. The > CANCEL request and 481 response appear as follows on my FS console: > > > recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: > ------------------------------------------------------------------------ > CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport > From: "Steve" > >;tag=as7f6965ea > To: > > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 > From: "Steve" > >;tag=as7f6965ea > To: > >;tag=71m745HKHKyjc > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 > CSeq: 103 CANCEL > Content-Length: 0 > -------------------------------------- > > > > Thanks. - SW > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/eca4313e/attachment-0001.html From willbelair at yahoo.com Tue Mar 24 07:23:42 2009 From: willbelair at yahoo.com (Will Smith) Date: Tue, 24 Mar 2009 07:23:42 -0700 (PDT) Subject: [Freeswitch-users] Console Window In-Reply-To: Message-ID: <488035.39134.qm@web53604.mail.re2.yahoo.com> Thank you all for your help. I use : /usr/local/freeswitch/bin/fs_cli? to open a FS instance. But then cannot use "shutdown" command. Then I used 'fsctl shutdown asap'? (directed by Szymon). It works perfectly. ? Again, thank you. Have a great day you all. ? Will --- On Tue, 3/24/09, Leon de Rooij wrote: From: Leon de Rooij Subject: Re: [Freeswitch-users] Console Window To: freeswitch-users at lists.freeswitch.org Date: Tuesday, March 24, 2009, 6:47 AM Just make sure you have mod_event socket loaded in conf/autoload_modules/modules.conf.xml : ?? ? And have it configured in conf/autoload_modules/event_socket.conf.xml Then you can use bin/fs_cli to connect to running FS instance. I don't think you can reconnect to a process of which you have disconnected the terminal (without using screen).. regards, Leon? On Mar 24, 2009, at 2:35 PM, Will Smith wrote: No, I started FS with this: /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? ? thank you for help --- On?Tue, 3/24/09, Saeed Ahmed??wrote: From: Saeed Ahmed Subject: Re: [Freeswitch-users] Console Window To:?freeswitch-users at lists.freeswitch.org Date: Tuesday, March 24, 2009, 6:25 AM Did you started FS with ?nc option? with this option you can connect to FS using ./fs_cli OR? use?screen! From:?freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]?On Behalf Of?Will Smith Sent:?Tuesday, March 24, 2009 1:52 PM To:?freeswitch-users at lists.freeswitch.org Subject:?[Freeswitch-users] Console Window ? Hi, I closed the console window that ran FS without shutting down FS. I had to restart the server to get back to the console window. I just wonder if there is a way to bring that window up without restart server. In the file /log/freeswitch.pid, I found a number, is that a seesion id? ? Thank you. ? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/99f67145/attachment.html From diego.viola at gmail.com Tue Mar 24 07:34:00 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 24 Mar 2009 10:34:00 -0400 Subject: [Freeswitch-users] Console Window In-Reply-To: <488035.39134.qm@web53604.mail.re2.yahoo.com> References: <488035.39134.qm@web53604.mail.re2.yahoo.com> Message-ID: <86a32abc0903240734j46513520k2f7f9b9525c1e18a@mail.gmail.com> I put /usr/local/freeswitch/bin in my path, like this. export PATH=/usr/local/freeswitch/bin:$PATH on my ~/.bashrc, you can put it on your /etc/profile if you want it to be global. Then I just do `freeswitch -nc' when I need to start it, `fs_cli' to connect, and `fsctl shutdown asap' to shut it down, or `freeswitch -stop'. Hope that helps. Diego 2009/3/24 Will Smith : > Thank you all for your help. > I use : /usr/local/freeswitch/bin/fs_cli? to open a FS instance. But then > cannot use "shutdown" command. Then I used 'fsctl shutdown asap'? (directed > by Szymon). It works perfectly. > > Again, thank you. Have a great day you all. > > Will > > --- On Tue, 3/24/09, Leon de Rooij wrote: > > From: Leon de Rooij > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:47 AM > > Just make sure you have mod_event socket loaded in > conf/autoload_modules/modules.conf.xml : > ?? ? > And have it configured in conf/autoload_modules/event_socket.conf.xml > Then you can use bin/fs_cli to connect to running FS instance. > I don't think you can reconnect to a process of which you have disconnected > the terminal (without using screen).. > regards, > Leon > On Mar 24, 2009, at 2:35 PM, Will Smith wrote: > > No, I started FS with this: > /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? > > thank you for help > > --- On?Tue, 3/24/09, Saeed Ahmed??wrote: > > From: Saeed Ahmed > Subject: Re: [Freeswitch-users] Console Window > To:?freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:25 AM > > Did you started FS with ?nc option? > with this option you can connect to FS using ./fs_cli > OR > use?screen! > > ________________________________ > > From:?freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org]?On Behalf Of?Will > Smith > Sent:?Tuesday, March 24, 2009 1:52 PM > To:?freeswitch-users at lists.freeswitch.org > Subject:?[Freeswitch-users] Console Window > > > > Hi, > > I closed the console window that ran FS without shutting down FS. I had to > restart the server to get back to the console window. I just wonder if there > is a way to bring that window up without restart server. In the file > /log/freeswitch.pid, I found a number, is that a seesion id? > > > > Thank you. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From solko at gcdf.pl Tue Mar 24 07:37:07 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 24 Mar 2009 15:37:07 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <488035.39134.qm@web53604.mail.re2.yahoo.com> References: <488035.39134.qm@web53604.mail.re2.yahoo.com> Message-ID: <49C8F013.702@gcdf.pl> Will Smith pisze: > Thank you all for your help. > I use : /usr/local/freeswitch/bin/fs_cli to open a FS instance. But > then cannot use "shutdown" command. Then I used 'fsctl shutdown asap' > (directed by Szymon). It works perfectly. > > Again, thank you. Have a great day you all. > Remeber there are other flags to shutdown freeswitch freeswitch at vertux> fsctl API CALL [fsctl()] output: -USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap|restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration [num]|loglevel [level]] Look for descriptions in wiki. > Will > > --- On *Tue, 3/24/09, Leon de Rooij //* wrote: > > From: Leon de Rooij > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:47 AM > > Just make sure you have mod_event socket loaded in > conf/autoload_modules/modules.conf.xml : > > > > And have it configured in conf/autoload_modules/event_socket.conf.xml > > Then you can use bin/fs_cli to connect to running FS instance. > > I don't think you can reconnect to a process of which you have > disconnected the terminal (without using screen).. > > regards, > > Leon > > On Mar 24, 2009, at 2:35 PM, Will Smith wrote: > >> No, I started FS with this: >> /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? >> >> thank you for help >> >> --- On *Tue, 3/24/09, Saeed Ahmed /> >/* wrote: >> >> From: Saeed Ahmed > > >> Subject: Re: [Freeswitch-users] Console Window >> To: freeswitch-users at lists.freeswitch.org >> >> Date: Tuesday, March 24, 2009, 6:25 AM >> >> Did you started FS with ?nc option? >> with this option you can connect to FS using ./fs_cli >> OR >> use /screen/! >> >> ------------------------------------------------------------------------ >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On >> Behalf Of *Will Smith >> *Sent:* Tuesday, March 24, 2009 1:52 PM >> *To:* freeswitch-users at lists.freeswitch.org >> >> *Subject:* [Freeswitch-users] Console Window >> >> >> >> Hi, >> >> I closed the console window that ran FS without shutting down >> FS. I had to restart the server to get back to the console >> window. I just wonder if there is a way to bring that window >> up without restart server. In the file /log/freeswitch.pid, I >> found a number, is that a seesion id? >> >> >> >> Thank you. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From grevenx at me.com Tue Mar 24 07:42:44 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Tue, 24 Mar 2009 15:42:44 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <488035.39134.qm@web53604.mail.re2.yahoo.com> References: <488035.39134.qm@web53604.mail.re2.yahoo.com> Message-ID: Please see documentation at: http://wiki.freeswitch.org/wiki/Fs_cli and http://wiki.freeswitch.org/wiki/Mod_commands Best regards, Even Andr? On 24. mars. 2009, at 15.23, Will Smith wrote: > Thank you all for your help. > I use : /usr/local/freeswitch/bin/fs_cli to open a FS instance. But > then cannot use "shutdown" command. Then I used 'fsctl shutdown > asap' (directed by Szymon). It works perfectly. > > Again, thank you. Have a great day you all. > > Will > > --- On Tue, 3/24/09, Leon de Rooij wrote: > From: Leon de Rooij > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:47 AM > > Just make sure you have mod_event socket loaded in conf/ > autoload_modules/modules.conf.xml : > > > > And have it configured in conf/autoload_modules/event_socket.conf.xml > > Then you can use bin/fs_cli to connect to running FS instance. > > I don't think you can reconnect to a process of which you have > disconnected the terminal (without using screen).. > > regards, > > Leon > > On Mar 24, 2009, at 2:35 PM, Will Smith wrote: > >> No, I started FS with this: >> /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? >> >> thank you for help >> >> --- On Tue, 3/24/09, Saeed Ahmed wrote: >> From: Saeed Ahmed >> Subject: Re: [Freeswitch-users] Console Window >> To: freeswitch-users at lists.freeswitch.org >> Date: Tuesday, March 24, 2009, 6:25 AM >> >> Did you started FS with ?nc option? >> with this option you can connect to FS using ./fs_cli >> OR >> use screen! >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Will Smith >> Sent: Tuesday, March 24, 2009 1:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Console Window >> >> >> Hi, >> >> I closed the console window that ran FS without shutting down FS. I >> had to restart the server to get back to the console window. I just >> wonder if there is a way to bring that window up without restart >> server. In the file /log/freeswitch.pid, I found a number, is that >> a seesion id? >> >> >> >> Thank you. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/5339773a/attachment-0001.html From dantavious at comcast.net Tue Mar 24 06:55:47 2009 From: dantavious at comcast.net (Derrick Edwards) Date: Tue, 24 Mar 2009 09:55:47 -0400 Subject: [Freeswitch-users] Sip for Skype - g.729 requirement In-Reply-To: <49C8D7EF.8050106@gnutelephony.org> References: <13257395.221237842869198.JavaMail.daniel@radio> <49C8D7EF.8050106@gnutelephony.org> Message-ID: <1237902947.28021.31.camel@Baby-Girls-Thang> It seems to me that they are just offering SIP trunking like voicepulse. Very interested to see what there rates would be like since they have a large infrastructure. On Tue, 2009-03-24 at 08:54 -0400, David Sugar wrote: > They require one use g.729, which is patent encumbered as well as rather > computationally intensive. > > Dan wrote: > > You probably already saw this but.... > > > > http://www.skypeforsip.com/ > > > > Skype is supporting sip for business users. > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Mar 24 07:54:34 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Mar 2009 10:54:34 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> Message-ID: <32586807-0E8B-4842-976C-97564F67823B@jerris.com> I note that its missing the to tag from the 180 sent 5 seconds earlier (I think thats okay) but the via branch tag is also different, which seems wrong. Can anyone else chime in, I can't recall the dialog matching rules of early dialog like this. Mike On Mar 24, 2009, at 9:57 AM, Steven Ward wrote: > Here it is: > > freeswitch at b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 at > 13:53:07.644865: > > ------------------------------------------------------------------------ > OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport > From: "Unknown" ;tag=as5adee8f4 > To: > Contact: > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Tue, 24 Mar 2009 13:53:07 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060 > From: "Unknown" ;tag=as5adee8f4 > To: ;tag=DytraHp3K84aD > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 > CSeq: 102 OPTIONS > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169: > > ------------------------------------------------------------------------ > INVITE sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport > From: "Steve" ;tag=as4863e49a > To: > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Tue, 24 Mar 2009 13:53:11 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 258 > v=0 > o=root 4756 4756 IN IP4 10.1.21.44 > s=session > c=IN IP4 10.1.21.44 > t=0 0 > m=audio 17956 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > ------------------------------------------------------------------------ > send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 > From: "Steve" ;tag=as4863e49a > To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Content-Length: 0 > > ------------------------------------------------------------------------ > send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660: > > ------------------------------------------------------------------------ > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 > From: "Steve" ;tag=as4863e49a > To: ;tag=e7KHcc76gHUXr > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > Proxy-Authenticate: Digest realm="10.1.21.44", > nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, > qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103: > > ------------------------------------------------------------------------ > ACK sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport > From: "Steve" ;tag=as4863e49a > To: ;tag=e7KHcc76gHUXr > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306: > > ------------------------------------------------------------------------ > INVITE sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport > From: "Steve" ;tag=as4863e49a > To: > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="b-pbx-lab-1", > realm="10.1.21.44", algorithm=MD5, uri="sip:70904 at b-pbx-lab-1.mynet.net > ", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", > response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, > cnonce="0e89cc90", nc=00000001 > Date: Tue, 24 Mar 2009 13:53:11 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 258 > v=0 > o=root 4756 4757 IN IP4 10.1.21.44 > s=session > c=IN IP4 10.1.21.44 > t=0 0 > m=audio 17956 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > ------------------------------------------------------------------------ > send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 > From: "Steve" ;tag=as4863e49a > To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/ > 70904 at 10.1.21.44 [1d28557e-187b-11de-8c60-ad87768304bc] > 2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing Steve->70904 in context default > 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes > [1d3a376c-187b-11de-8c60-ad87768304bc] > send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291: > > ------------------------------------------------------------------------ > INVITE sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c SIP/ > 2.0 > Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p > Max-Forwards: 69 > From: "Steve" ;tag=gS62F28DB372F > To: > Call-ID: f4992499-931d-122c-34b1-003018ae1862 > CSeq: 112833059 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 328 > Remote-Party-ID: "Steve" 70904 at 10.1.21.45>;screen=yes;privacy=off > v=0 > o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 > 10.1.21.45 > s=FreeSWITCH > c=IN IP4 10.1.21.45 > t=0 0 > m=audio 22432 RTP/AVP 0 9 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589: > > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p > Contact: > To: 70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>;tag=fa138551 > From: "Steve";tag=gS62F28DB372F > Call-ID: f4992499-931d-122c-34b1-003018ae1862 > CSeq: 112833059 INVITE > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes > ! > send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060: > > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 > From: "Steve" ;tag=as4863e49a > To: ;tag=FgDae7QaetHgm > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 > sofia_receive_message() Ring-Ready sofia/internal/70904 at 10.1.21.44! > 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 > switch_ivr_originate() Ring Ready sofia/internal/70904 at 10.1.21.44! > recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013: > > ------------------------------------------------------------------------ > CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport > From: "Steve" ;tag=as4863e49a > To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618: > > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060 > From: "Steve" ;tag=as4863e49a > To: ;tag=FgDae7QaetHgm > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 CANCEL > Content-Length: 0 > > ------------------------------------------------------------------------ > > > > 2009/3/24 Michael Jerris > This means we could not match the cancel to a current call dialog. > I would need to see the full sip trace of the call to know why, but > typically this is because of not matching call Id or to or from tags > > Mike > > > On Mar 24, 2009, at 9:43 AM, Steven Ward > wrote: > >> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b- >> lab-1) while the call is still ringing does not work. >> >> Why is this request resulting in a 481? >> >> I appreciate the help - I'm still just starting to learn SIP & FS. >> The CANCEL request and 481 response appear as follows on my FS >> console: >> >> >> recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: >> >> ------------------------------------------------------------------------ >> CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport >> From: "Steve" ;tag=as7f6965ea >> To: >> Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >> CSeq: 103 CANCEL >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: >> >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 >> From: "Steve" ;tag=as7f6965ea >> To: ;tag=71m745HKHKyjc >> Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >> CSeq: 103 CANCEL >> Content-Length: 0 >> >> -------------------------------------- >> >> >> >> Thanks. - SW >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/fd0f983a/attachment-0001.html From leon at scarlet-internet.nl Tue Mar 24 07:55:37 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 24 Mar 2009 15:55:37 +0100 Subject: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user Message-ID: Hi, I'm trying to get some cli commands working in combination with xml- curl. Endpoints are parsed properly for SIP registrations and invites, but when I use the CLI command "user_exists" it returns false, while I do return an endpoint (same syntax as for a sofia_reg_parse_auth event) on the webserver. Should switch_xml_locate_user event receive a different syntax ? freeswitch at internal> user_exists accountcode gigaset toyos.nl false XML returned from webserver:
thanks & regards, Leon From mike at jerris.com Tue Mar 24 07:59:45 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Mar 2009 10:59:45 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <32586807-0E8B-4842-976C-97564F67823B@jerris.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> <32586807-0E8B-4842-976C-97564F67823B@jerris.com> Message-ID: <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> This appears to be a bug in FreeSWITCH. Can you please test this on current svn trunk and if it is still a problem, please report this as a bug to http://jira.freeswitch.org. MIke On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote: > I note that its missing the to tag from the 180 sent 5 seconds > earlier (I think thats okay) but the via branch tag is also > different, which seems wrong. Can anyone else chime in, I can't > recall the dialog matching rules of early dialog like this. > > Mike > > On Mar 24, 2009, at 9:57 AM, Steven Ward wrote: > >> Here it is: >> >> freeswitch at b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 >> at 13:53:07.644865: >> >> ------------------------------------------------------------------------ >> OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport >> From: "Unknown" ;tag=as5adee8f4 >> To: >> Contact: >> Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Date: Tue, 24 Mar 2009 13:53:07 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Supported: replaces >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060 >> From: "Unknown" ;tag=as5adee8f4 >> To: ;tag=DytraHp3K84aD >> Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 >> CSeq: 102 OPTIONS >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: 100rel, timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, include- >> session-description, presence.winfo, message-summary, refer >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169: >> >> ------------------------------------------------------------------------ >> INVITE sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport >> From: "Steve" ;tag=as4863e49a >> To: >> Contact: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Date: Tue, 24 Mar 2009 13:53:11 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Supported: replaces >> Content-Type: application/sdp >> Content-Length: 258 >> v=0 >> o=root 4756 4756 IN IP4 10.1.21.44 >> s=session >> c=IN IP4 10.1.21.44 >> t=0 0 >> m=audio 17956 RTP/AVP 0 8 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> ------------------------------------------------------------------------ >> send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 102 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660: >> >> ------------------------------------------------------------------------ >> SIP/2.0 407 Proxy Authentication Required >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: ;tag=e7KHcc76gHUXr >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 102 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: 100rel, timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, include- >> session-description, presence.winfo, message-summary, refer >> Proxy-Authenticate: Digest realm="10.1.21.44", >> nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, >> qop="auth" >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103: >> >> ------------------------------------------------------------------------ >> ACK sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport >> From: "Steve" ;tag=as4863e49a >> To: ;tag=e7KHcc76gHUXr >> Contact: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 102 ACK >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306: >> >> ------------------------------------------------------------------------ >> INVITE sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport >> From: "Steve" ;tag=as4863e49a >> To: >> Contact: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 INVITE >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Proxy-Authorization: Digest username="b-pbx-lab-1", >> realm="10.1.21.44", algorithm=MD5, uri="sip:70904 at b-pbx-lab-1.mynet.net >> ", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", >> response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, >> cnonce="0e89cc90", nc=00000001 >> Date: Tue, 24 Mar 2009 13:53:11 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Supported: replaces >> Content-Type: application/sdp >> Content-Length: 258 >> v=0 >> o=root 4756 4757 IN IP4 10.1.21.44 >> s=session >> c=IN IP4 10.1.21.44 >> t=0 0 >> m=audio 17956 RTP/AVP 0 8 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> ------------------------------------------------------------------------ >> send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 >> switch_channel_set_name() New Channel sofia/internal/ >> 70904 at 10.1.21.44 [1d28557e-187b-11de-8c60-ad87768304bc] >> 2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing Steve->70904 in context default >> 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 >> switch_channel_set_name() New Channel sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes >> [1d3a376c-187b-11de-8c60-ad87768304bc] >> send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291: >> >> ------------------------------------------------------------------------ >> INVITE sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c >> SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p >> Max-Forwards: 69 >> From: "Steve" ;tag=gS62F28DB372F >> To: >> Call-ID: f4992499-931d-122c-34b1-003018ae1862 >> CSeq: 112833059 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: 100rel, timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, include- >> session-description, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 328 >> Remote-Party-ID: "Steve" > 70904 at 10.1.21.45>;screen=yes;privacy=off >> v=0 >> o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 >> 10.1.21.45 >> s=FreeSWITCH >> c=IN IP4 10.1.21.45 >> t=0 0 >> m=audio 22432 RTP/AVP 0 9 8 3 101 13 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589: >> >> ------------------------------------------------------------------------ >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p >> Contact: >> To: > 70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>;tag=fa138551 >> From: "Steve";tag=gS62F28DB372F >> Call-ID: f4992499-931d-122c-34b1-003018ae1862 >> CSeq: 112833059 INVITE >> User-Agent: X-Lite release 1011s stamp 41150 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-03-24 09:53:11 [NOTICE] sofia.c:2782 >> sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes >> ! >> send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060: >> >> ------------------------------------------------------------------------ >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: ;tag=FgDae7QaetHgm >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: 100rel, timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, include- >> session-description, presence.winfo, message-summary, refer >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 >> sofia_receive_message() Ring-Ready sofia/internal/70904 at 10.1.21.44! >> 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 >> switch_ivr_originate() Ring Ready sofia/internal/70904 at 10.1.21.44! >> recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013: >> >> ------------------------------------------------------------------------ >> CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport >> From: "Steve" ;tag=as4863e49a >> To: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 CANCEL >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618: >> >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: ;tag=FgDae7QaetHgm >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 CANCEL >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> >> >> >> 2009/3/24 Michael Jerris >> This means we could not match the cancel to a current call dialog. >> I would need to see the full sip trace of the call to know why, but >> typically this is because of not matching call Id or to or from tags >> >> Mike >> >> >> On Mar 24, 2009, at 9:43 AM, Steven Ward >> wrote: >> >>> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b- >>> lab-1) while the call is still ringing does not work. >>> >>> Why is this request resulting in a 481? >>> >>> I appreciate the help - I'm still just starting to learn SIP & >>> FS. The CANCEL request and 481 response appear as follows on my >>> FS console: >>> >>> >>> recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: >>> >>> ------------------------------------------------------------------------ >>> CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >>> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport >>> From: "Steve" ;tag=as7f6965ea >>> To: >>> Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >>> CSeq: 103 CANCEL >>> User-Agent: Asterisk PBX >>> Max-Forwards: 70 >>> Content-Length: 0 >>> >>> ------------------------------------------------------------------------ >>> send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 481 Call/Transaction Does Not Exist >>> Via: SIP/2.0/UDP >>> 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 >>> From: "Steve" ;tag=as7f6965ea >>> To: ;tag=71m745HKHKyjc >>> Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >>> CSeq: 103 CANCEL >>> Content-Length: 0 >>> >>> -------------------------------------- >>> >>> >>> >>> Thanks. - SW >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/ecceb34b/attachment-0001.html From steve.d.ward at gmail.com Tue Mar 24 08:06:15 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 11:06:15 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> <32586807-0E8B-4842-976C-97564F67823B@jerris.com> <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> Message-ID: <4ea6e8f20903240806p45016e6fs90e44512437a8060@mail.gmail.com> Mike, Thanks for taking the time to look at this - I appreciate it. I'll go ahead and test it out on the current svn trunk. - Steve 2009/3/24 Michael Jerris > This appears to be a bug in FreeSWITCH. Can you please test this on > current svn trunk and if it is still a problem, please report this as a bug > to http://jira.freeswitch.org. > MIke > > On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote: > > I note that its missing the to tag from the 180 sent 5 seconds earlier (I > think thats okay) but the via branch tag is also different, which seems > wrong. Can anyone else chime in, I can't recall the dialog matching rules > of early dialog like this. > Mike > > On Mar 24, 2009, at 9:57 AM, Steven Ward wrote: > > Here it is: > > freeswitch at b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 at > 13:53:07.644865: > ------------------------------------------------------------------------ > OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport > From: "Unknown" > >;tag=as5adee8f4 > To: > Contact: > > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Tue, 24 Mar 2009 13:53:07 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > ------------------------------------------------------------------------ > send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060 > From: "Unknown" > >;tag=as5adee8f4 > To: ;tag=DytraHp3K84aD > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 > CSeq: 102 OPTIONS > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Length: 0 > ------------------------------------------------------------------------ > recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169: > ------------------------------------------------------------------------ > INVITE sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport > From: "Steve" > >;tag=as4863e49a > To: > > > Contact: > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Tue, 24 Mar 2009 13:53:11 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 258 > v=0 > o=root 4756 4756 IN IP4 10.1.21.44 > s=session > c=IN IP4 10.1.21.44 > t=0 0 > m=audio 17956 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Content-Length: 0 > ------------------------------------------------------------------------ > send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660: > ------------------------------------------------------------------------ > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > >;tag=e7KHcc76gHUXr > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Proxy-Authenticate: Digest realm="10.1.21.44", > nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth" > Content-Length: 0 > ------------------------------------------------------------------------ > recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103: > ------------------------------------------------------------------------ > ACK sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport > From: "Steve" > >;tag=as4863e49a > To: > >;tag=e7KHcc76gHUXr > Contact: > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > ------------------------------------------------------------------------ > recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306: > ------------------------------------------------------------------------ > INVITE sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport > From: "Steve" > >;tag=as4863e49a > To: > > > Contact: > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="b-pbx-lab-1", realm="10.1.21.44", > algorithm=MD5, uri="sip:70904 at b-pbx-lab-1.mynet.net", > nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", > response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, cnonce="0e89cc90", > nc=00000001 > Date: Tue, 24 Mar 2009 13:53:11 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 258 > v=0 > o=root 4756 4757 IN IP4 10.1.21.44 > s=session > c=IN IP4 10.1.21.44 > t=0 0 > m=audio 17956 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() > New Channel sofia/internal/70904 at 10.1.21.44[1d28557e-187b-11de-8c60-ad87768304bc] > 2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing Steve->70904 in context default > 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() > New Channel sofia/internal/ > sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes[1d3a376c-187b-11de-8c60-ad87768304bc] > send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291: > ------------------------------------------------------------------------ > INVITE sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p > Max-Forwards: 69 > From: "Steve" > >;tag=gS62F28DB372F > To: > Call-ID: f4992499-931d-122c-34b1-003018ae1862 > CSeq: 112833059 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 328 > Remote-Party-ID: "Steve" > >;screen=yes;privacy=off > v=0 > o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 10.1.21.45 > s=FreeSWITCH > c=IN IP4 10.1.21.45 > t=0 0 > m=audio 22432 RTP/AVP 0 9 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > ------------------------------------------------------------------------ > recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p > Contact: > To: >;tag=fa138551 > From: "Steve" > >;tag=gS62F28DB372F > Call-ID: f4992499-931d-122c-34b1-003018ae1862 > CSeq: 112833059 INVITE > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/ > sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes! > send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > >;tag=FgDae7QaetHgm > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 sofia_receive_message() > Ring-Ready sofia/internal/70904 at 10.1.21.44! > 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 > switch_ivr_originate() Ring Ready sofia/internal/70904 at 10.1.21.44! > recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013: > ------------------------------------------------------------------------ > CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport > From: "Steve" > >;tag=as4863e49a > To: > > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > ------------------------------------------------------------------------ > send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618: > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > >;tag=FgDae7QaetHgm > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 CANCEL > Content-Length: 0 > ------------------------------------------------------------------------ > > > > 2009/3/24 Michael Jerris > >> This means we could not match the cancel to a current call dialog. I >> would need to see the full sip trace of the call to know why, but typically >> this is because of not matching call Id or to or from tags >> >> Mike >> >> >> On Mar 24, 2009, at 9:43 AM, Steven Ward wrote: >> >> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) >> while the call is still ringing does not work. >> >> Why is this request resulting in a 481? >> >> I appreciate the help - I'm still just starting to learn SIP & FS. The >> CANCEL request and 481 response appear as follows on my FS console: >> >> >> recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: >> >> ------------------------------------------------------------------------ >> CANCEL sip: >> 70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport >> From: "Steve" 70904 at 10.1.21.44 >> >;tag=as7f6965ea >> To: 70904 at b-lab-1.mynet.net> >> Call-ID: <237598fd102b739a03b4a4047bf69843 at 10.1.21.44> >> 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >> CSeq: 103 CANCEL >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: >> >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 >> From: "Steve" 70904 at 10.1.21.44 >> >;tag=as7f6965ea >> To: 70904 at b-lab-1.mynet.net >> >;tag=71m745HKHKyjc >> Call-ID: <237598fd102b739a03b4a4047bf69843 at 10.1.21.44> >> 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >> CSeq: 103 CANCEL >> Content-Length: 0 >> -------------------------------------- >> >> >> >> Thanks. - SW >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/5ecb5fd8/attachment-0001.html From woof at nortel.com Tue Mar 24 09:32:16 2009 From: woof at nortel.com (Andy Spitzer) Date: Tue, 24 Mar 2009 12:32:16 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <4ea6e8f20903240806p45016e6fs90e44512437a8060@mail.gmail.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> <32586807-0E8B-4842-976C-97564F67823B@jerris.com> <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> <4ea6e8f20903240806p45016e6fs90e44512437a8060@mail.gmail.com> Message-ID: Woof! Appears to be a recently fixed * bug: 0014431: Bad branch parameter value in CANCEL request http://bugs.digium.com/view.php?id=14431 --Woof! From dujinfang at gmail.com Tue Mar 24 10:30:57 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 25 Mar 2009 01:30:57 +0800 Subject: [Freeswitch-users] FreeSWITCH Chinese Community Message-ID: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> Hi all, In about a year of playing with FreeSWITCH, I really like it. Not only the software is great but also the community. As people keep asking me where can find some documentation in Chinese, I told them if they want to play in deep they need to know English better. However, the fact is, even one can read English without problem, he still prefer find some information in his native language first. So, I'd like to start a Chinese community hoping FreeSWITCH can get more popular in China. As we saw big telecom companies merged in the last year, I bet VoIP technology and application will go more and more faster in the near future... While FS official site has plenty of documentation, obviously we don't want to translate word by word. Then where should we start from? A forum? There was a big argue between a forum and mailing list in the last few weeks, and finally an English forum and an Italian one is out. While I can find server in China, the major pain is that running any kind of BBS in China mainland need to get some kind of permissions by the government first. Any idea, suggestion? Anyone want to help or cooperate about this? And, if I can get this runing, can I use the FreeSWITCH logo and www.freeswitch.org.cn ? Best regards, Seven. From msc at freeswitch.org Tue Mar 24 11:03:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Mar 2009 11:03:51 -0700 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> Message-ID: <87f2f3b90903241103l30302179u655713047b0d4bf9@mail.gmail.com> > While FS official site has plenty of documentation, obviously we don't > want to translate word by word. ?Then where should we start from? A > forum? I would start by finding as many people as possible who are literate in both English and Chinese and who are willing to help out. Once you have your group assembled you will have a better idea of what your goals should be. More people helping will make things easier to accomplish. There was a big argue between a forum and mailing list in the > last few weeks, and finally an English forum and an Italian one is > out. ?While I can find server in China, the major pain is that running > any kind of BBS in China mainland need to get some kind of permissions > by the government first. > > Any idea, suggestion? Anyone want to help or cooperate about this? I am looking into creating a multi-language wiki at wiki.freeswitch.org. Anyone with experience in setting up multiple languages with MediaWiki software please contact me. So far we have people willing to help create documentation in French, Spanish, Portugese, Russian, and now Chinese. I think we have some Italians out there as well! (ciao bella) Please email me off list if you are in a position to assist with the wiki and languages other than English. > > And, if I can get this runing, can I use the FreeSWITCH logo and www.freeswitch.org.cn > ?? Please direct questions about the FreeSWITCH logo and domain names to consulting at freeswitch.org. FreeSWITCH and the logo are trademarks so it would be best to get permission from Anthony before doing anything. -MC From mitul at enterux.com Tue Mar 24 11:32:20 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 25 Mar 2009 00:02:20 +0530 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> Message-ID: Hello, I can provide you with the hosting on our box in US let me know. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 24-Mar-09, at 23:00, dujinfang wrote: > Hi all, > > In about a year of playing with FreeSWITCH, I really like it. Not only > the software is great but also the community. As people keep asking me > where can find some documentation in Chinese, I told them if they want > to play in deep they need to know English better. However, the fact > is, even one can read English without problem, he still prefer find > some information in his native language first. So, I'd like to start a > Chinese community hoping FreeSWITCH can get more popular in China. As > we saw big telecom companies merged in the last year, I bet VoIP > technology and application will go more and more faster in the near > future... > > While FS official site has plenty of documentation, obviously we don't > want to translate word by word. Then where should we start from? A > forum? There was a big argue between a forum and mailing list in the > last few weeks, and finally an English forum and an Italian one is > out. While I can find server in China, the major pain is that running > any kind of BBS in China mainland need to get some kind of permissions > by the government first. > > Any idea, suggestion? Anyone want to help or cooperate about this? > > And, if I can get this runing, can I use the FreeSWITCH logo and www.freeswitch.org.cn > ? > > Best regards, > Seven. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Mar 24 11:42:33 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Mar 2009 13:42:33 -0500 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> Message-ID: <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> How about we all work together and work on the FreeSWITCH.org infrastructure instead of spreading the resources thinner and thinner till nobody is doing really much of anything. We need people to step up and help out with the website, wiki, jira, irc, testing and various other things that currently are spread thin. ;) Please!!! I'm not asking everyone to step up and code in C... /b On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote: > Hello, > > I can provide you with the hosting on our box in US let me know. > > Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt Ltd, > The Enterprise Linux Company(r), > http://www.enterux.com/ > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/d846c84c/attachment.html From steve.d.ward at gmail.com Tue Mar 24 11:45:01 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 14:45:01 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> <32586807-0E8B-4842-976C-97564F67823B@jerris.com> <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> <4ea6e8f20903240806p45016e6fs90e44512437a8060@mail.gmail.com> Message-ID: <4ea6e8f20903241145t70aeb0asd37376429367974a@mail.gmail.com> Ah ha! Thanks for finding that. I updated my * server and I'm all set. Many thanks for all the feedback and help. 2009/3/24 Andy Spitzer > Woof! > > Appears to be a recently fixed * bug: > > 0014431: Bad branch parameter value in CANCEL request > http://bugs.digium.com/view.php?id=14431 > > --Woof! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/742e974c/attachment.html From mitul at enterux.com Tue Mar 24 11:56:09 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 25 Mar 2009 00:26:09 +0530 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> Message-ID: <9DBFC7A7-DA73-4CF2-AC91-4BF3EF914884@enterux.com> Brian, I can help with website, wiki and testing, tell me what's next step forward. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 25-Mar-09, at 0:12, Brian West wrote: > How about we all work together and work on the FreeSWITCH.org > infrastructure instead of spreading the resources thinner and > thinner till nobody is doing really much of anything. > > We need people to step up and help out with the website, wiki, jira, > irc, testing and various other things that currently are spread > thin. ;) Please!!! I'm not asking everyone to step up and code in > C... > > /b > > On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote: > >> Hello, >> >> I can provide you with the hosting on our box in US let me know. >> >> Regards, >> Mitul Limbani, >> Founder & CEO, >> Enterux Solutions Pvt Ltd, >> The Enterprise Linux Company(r), >> http://www.enterux.com/ >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/b6d78784/attachment.html From msc at freeswitch.org Tue Mar 24 12:05:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Mar 2009 12:05:21 -0700 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <9DBFC7A7-DA73-4CF2-AC91-4BF3EF914884@enterux.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> <9DBFC7A7-DA73-4CF2-AC91-4BF3EF914884@enterux.com> Message-ID: <87f2f3b90903241205g72570c3w92c99317d2c86fc5@mail.gmail.com> 2009/3/24 Mitul Limbani : > Brian, > I can help with website, wiki and testing, tell me what's next step forward. > Mitul, do you have any experience with MediaWiki? -MC From dujinfang at gmail.com Tue Mar 24 17:42:44 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 25 Mar 2009 08:42:44 +0800 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <87f2f3b90903241103l30302179u655713047b0d4bf9@mail.gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <87f2f3b90903241103l30302179u655713047b0d4bf9@mail.gmail.com> Message-ID: <0C9F4BE9-F6AE-444B-993C-FF093DBA9CE0@gmail.com> On Mar 25, 2009, at 2:03 AM, Michael Collins wrote: >> While FS official site has plenty of documentation, obviously we >> don't >> want to translate word by word. Then where should we start from? A >> forum? > > I would start by finding as many people as possible who are literate > in both English and Chinese and who are willing to help out. Once you > have your group assembled you will have a better idea of what your > goals should be. More people helping will make things easier to > accomplish. Ppl not only literate in both English and Chinse, but also need to has good knowledge of VoIP and FreeSWITCH. > There was a big argue between a forum and mailing list in the >> last few weeks, and finally an English forum and an Italian one is >> out. While I can find server in China, the major pain is that >> running >> any kind of BBS in China mainland need to get some kind of >> permissions >> by the government first. >> >> Any idea, suggestion? Anyone want to help or cooperate about this? > > I am looking into creating a multi-language wiki at > wiki.freeswitch.org. Anyone with experience in setting up multiple > languages with MediaWiki software please contact me. So far we have > people willing to help create documentation in French, Spanish, > Portugese, Russian, and now Chinese. I think we have some Italians out > there as well! (ciao bella) > > Please email me off list if you are in a position to assist with the > wiki and languages other than English. > How's the plan of a multi-language wiki? I think mediaWiKi do support multi-language. So, maybe we can start from wiki.freeswitch.org/zh_CN/ or some structure else. While it is possible to follow the change log of the EN wiki to change Chinese version accordingly, it would be challenge to make sure the consistency. And regardlessly what if Chinese version change first? I'd like to help with this. But I cannot translate all hundreds of pages and make it consistency by my self. But I'm pretty sure that we would get a group of ppl to do this. So the start maybe either translate some key pages and make the consistency, or translate as much pages ... Email me off list on how to start this. :) >> >> And, if I can get this runing, can I use the FreeSWITCH logo and www.freeswitch.org.cn >> ? > > Please direct questions about the FreeSWITCH logo and domain names to > consulting at freeswitch.org. FreeSWITCH and the logo are trademarks so > it would be best to get permission from Anthony before doing anything. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Tue Mar 24 17:47:02 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 25 Mar 2009 08:47:02 +0800 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> Message-ID: <63686540-6117-4B42-AF21-25A5659DC959@gmail.com> On Mar 25, 2009, at 2:42 AM, Brian West wrote: > How about we all work together and work on the FreeSWITCH.org > infrastructure instead of spreading the resources thinner and > thinner till nobody is doing really much of anything. > I agree with this and would like to work with the wiki.freeswitch.org on the EN and CN language part. However, I don't think we can maintain a Chinese version of IRC, seems still we need a Chinese BBS. > We need people to step up and help out with the website, wiki, jira, > irc, testing and various other things that currently are spread > thin. ;) Please!!! I'm not asking everyone to step up and code in > C... > Take me in :) > /b > > On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote: > >> Hello, >> >> I can provide you with the hosting on our box in US let me know. >> >> Regards, >> Mitul Limbani, >> Founder & CEO, >> Enterux Solutions Pvt Ltd, >> The Enterprise Linux Company(r), >> http://www.enterux.com/ >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/d78c79c3/attachment.html From brian at freeswitch.org Tue Mar 24 17:53:53 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Mar 2009 19:53:53 -0500 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <63686540-6117-4B42-AF21-25A5659DC959@gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> <63686540-6117-4B42-AF21-25A5659DC959@gmail.com> Message-ID: <4F0F01A9-39DC-4234-8301-6996F9BC9255@freeswitch.org> On Mar 24, 2009, at 7:47 PM, dujinfang wrote: > > On Mar 25, 2009, at 2:42 AM, Brian West wrote: >> How about we all work together and work on the FreeSWITCH.org >> infrastructure instead of spreading the resources thinner and >> thinner till nobody is doing really much of anything. >> > > I agree with this and would like to work with the > wiki.freeswitch.org on the EN and CN language part. However, I don't > think we can maintain a Chinese version of IRC, seems still we need > a Chinese BBS. Someone tried to do a spanish IRC channel it doesn't have anyone in it but me most of the time... hrm...... I'm not trying to discount anything here but it seems language specific anything seems to not take off. We need more cheerleaders.... btw cluecon is coming up.... remember to register early! ;) >> We need people to step up and help out with the website, wiki, >> jira, irc, testing and various other things that currently are >> spread thin. ;) Please!!! I'm not asking everyone to step up and >> code in C... >> > > Take me in :) What do you want to be involved in? #freeswitch-web is the correct channel to be in if you want to help out on IRC. ;) > >> /b From msc at freeswitch.org Tue Mar 24 18:01:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Mar 2009 18:01:44 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: ClueCon 2009 - Register Today! Message-ID: <87f2f3b90903241801q71b33a80n1cbf5cbd525e86ae@mail.gmail.com> The FreeSWITCH Team is pleased to announce that all may register for ClueCon 2009 immediately! ClueCon is the Telephony Developers Conference by developers, for developers. This year's event will be held at the beautiful Wyndham Hotel in Chicago, August 4-6. Special room rates have been secured and you will enjoy staying at this luxurious hotel. The conference attendee price is only $499 per person, but hurry! This early bird special price is valid only until June 30th. The schedule is still being finalized, however you can expect to hear from notable names in the world of telephony. You will also have opportunities to spend one-on-one time with developers, vendors, and other attendees. One special highlight for this year is the great MacBook Pro giveaway! On Thursday at the end of the conference a 15" MacBook Pro will be given away to one lucky attendee. This laptop will be laser engraved with the logos of all of the sponsors for ClueCon 2009 and will be used as the presenters' laptop for the entire conference. Start making your travel plans right away. Visit http://www.cluecon.com and sign up today. We look forward to seeing you in Chicago this August! Michael S Collins 877-742-CLUE From mitul at enterux.com Tue Mar 24 19:30:25 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 25 Mar 2009 08:00:25 +0530 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <87f2f3b90903241205g72570c3w92c99317d2c86fc5@mail.gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> <9DBFC7A7-DA73-4CF2-AC91-4BF3EF914884@enterux.com> <87f2f3b90903241205g72570c3w92c99317d2c86fc5@mail.gmail.com> Message-ID: <74493056-0544-4833-BF9A-90CFBBCC90B4@enterux.com> MC, I don't have much exp with mediawiki but have some with drupal. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 25-Mar-09, at 0:35, Michael Collins wrote: > 2009/3/24 Mitul Limbani : >> Brian, >> I can help with website, wiki and testing, tell me what's next step >> forward. >> > > Mitul, do you have any experience with MediaWiki? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raul at etellicom.com Tue Mar 24 22:53:30 2009 From: raul at etellicom.com (Raul Fragoso) Date: Wed, 25 Mar 2009 02:53:30 -0300 Subject: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user In-Reply-To: References: Message-ID: <1237960410.18715.0.camel@raul-laptop> You should use the id value of the user, not accountcode: user_exists id gigaset toyos.nl Regards, Raul On Tue, 2009-03-24 at 15:55 +0100, Leon de Rooij wrote: > Hi, > > I'm trying to get some cli commands working in combination with xml- > curl. > > Endpoints are parsed properly for SIP registrations and invites, but > when I use the CLI command "user_exists" it returns false, while I do > return an endpoint (same syntax as for a sofia_reg_parse_auth event) > on the webserver. > > Should switch_xml_locate_user event receive a different syntax ? > > freeswitch at internal> user_exists accountcode gigaset toyos.nl > false > > XML returned from webserver: > > > >
> > > > value="a211336e29c4f3756e6af343ce6da27b"/> > > > > > > > > > > > > > > > > > >
> > > thanks & regards, > > Leon > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mrene_lists at avgs.ca Tue Mar 24 22:57:07 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 25 Mar 2009 01:57:07 -0400 Subject: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user In-Reply-To: <1237960410.18715.0.camel@raul-laptop> References: <1237960410.18715.0.camel@raul-laptop> Message-ID: As Raul said, the user_exists function will look for a user *attribute* so unless you write your user as , you need to use id (and the later wouldnt set the accountcode variable, no) Math On 25-Mar-09, at 1:53 AM, Raul Fragoso wrote: > You should use the id value of the user, not accountcode: > > user_exists id gigaset toyos.nl > > Regards, > > Raul > > On Tue, 2009-03-24 at 15:55 +0100, Leon de Rooij wrote: >> Hi, >> >> I'm trying to get some cli commands working in combination with xml- >> curl. >> >> Endpoints are parsed properly for SIP registrations and invites, but >> when I use the CLI command "user_exists" it returns false, while I do >> return an endpoint (same syntax as for a sofia_reg_parse_auth event) >> on the webserver. >> >> Should switch_xml_locate_user event receive a different syntax ? >> >> freeswitch at internal> user_exists accountcode gigaset toyos.nl >> false >> >> XML returned from webserver: >> >> >> >>
>> >> >> >> > value="a211336e29c4f3756e6af343ce6da27b"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> >> >> thanks & regards, >> >> Leon >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From codecomplete at free.fr Wed Mar 25 02:51:32 2009 From: codecomplete at free.fr (Gilles) Date: Wed, 25 Mar 2009 10:51:32 +0100 Subject: [Freeswitch-users] [Remote SIP client] Couple of questions Message-ID: <7.0.1.0.2.20090325104634.02701c88@fredshack.com> Hello, I have a couple of questions related to having SIP users connecting from the Net to a Freeswitch server through NAT routers on both ends: 1. How must I configure routers on both ends? I understand that I need to route incoming TCP/UDP 5080 into the Freeswitch server, but what about the other router? I guess I also need to route this port to let the SIP phone ring, but what about data (RTP/RTCP)? 2. The Freeswitch server is connected to the POTS with either an OpenVox PCI card or a Linksys 3102 box: When a call is made between the POTS and a remote SIP phone (ie. out there on the Net, not on the same LAN as the Freeswitch server), is there a way for data to flow directly from the POTS to the remote SIP client instead of through the Freeswitch server? Thank you. From leon at scarlet-internet.nl Wed Mar 25 03:15:01 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Wed, 25 Mar 2009 11:15:01 +0100 Subject: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user In-Reply-To: References: <1237960410.18715.0.camel@raul-laptop> Message-ID: <9B6C044E-B6EC-48A9-B711-0E3A3AD48142@scarlet-internet.nl> Thanks.. I see it's even documented at http://wiki.freeswitch.org/wiki/Mod_commands#user_exists (doh) regards, Leon On Mar 25, 2009, at 6:57 AM, Mathieu Rene wrote: > As Raul said, the user_exists function will look for a user > *attribute* so unless you write your user as accountcode="blah2">, you need to use id (and the later wouldnt set > the accountcode variable, no) > > Math > > On 25-Mar-09, at 1:53 AM, Raul Fragoso wrote: > >> You should use the id value of the user, not accountcode: >> >> user_exists id gigaset toyos.nl >> >> Regards, >> >> Raul >> >> On Tue, 2009-03-24 at 15:55 +0100, Leon de Rooij wrote: >>> Hi, >>> >>> I'm trying to get some cli commands working in combination with xml- >>> curl. >>> >>> Endpoints are parsed properly for SIP registrations and invites, but >>> when I use the CLI command "user_exists" it returns false, while I >>> do >>> return an endpoint (same syntax as for a sofia_reg_parse_auth >>> event) >>> on the webserver. >>> >>> Should switch_xml_locate_user event receive a different syntax ? >>> >>> freeswitch at internal> user_exists accountcode gigaset toyos.nl >>> false >>> >>> XML returned from webserver: >>> >>> >>> >>>
>>> >>> >>> >>> >> value="a211336e29c4f3756e6af343ce6da27b"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>>
>>> >>> >>> thanks & regards, >>> >>> Leon >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From leon at scarlet-internet.nl Wed Mar 25 03:16:54 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Wed, 25 Mar 2009 11:16:54 +0100 Subject: [Freeswitch-users] user_data sends two header_strings named key Message-ID: Hi, I think this is a bug in mod_commands.c : When using "user_data" function from mod_commands (mod_commands.c, line 358) A header string "key" is added to params (mod_commands.c, line 362) Then switch_xml_locate_user is called (switch_xml.c, line 1712) And that function adds another header string "key" (switch_xml.c, line 1740) This results in two keys with the same name being sent ( to in my case mod_xml_curl ). I opened an issue in jira for this (is it alright like that ?) : http://jira.freeswitch.org/browse/MODAPP-242 regards, Leon From codecomplete at free.fr Wed Mar 25 04:08:34 2009 From: codecomplete at free.fr (Gilles) Date: Wed, 25 Mar 2009 12:08:34 +0100 Subject: [Freeswitch-users] Starting Freeswitch at boot-time with rc.d script? Message-ID: <7.0.1.0.2.20090325120411.024773f0@fredshack.com> Hello I'm struggling to have Freeswitch start automatically in a Suse host at boot-time through the rc.d script provided in SVN. Here's what I did so far: 1. cp /usr/src/freeswitch/build/freeswitch.init.suse /etc/init.d/freeswitch 2. chmod 755 /etc/init.d/freeswitch 3. chkconfig freeswitch 345 4. chkconfig -l freeswitch 5. ln -s /usr/sbin/rcfreeswitch /etc/init.d/freeswitch 6. Edited /etc/init.d/freeswitch to correct the following two lines: #FREESWITCH_BIN=/opt/freeswitch/bin/freeswitch FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitch #FREESWITCH_CONFIG=/etc/sysconfig/freeswitch FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml 7. Ran the script: /etc/init.d/freeswitch start [...] Session Rate[30] SQL [Enabled] 2009-03-25 11:54:56 [WARNING] switch_console.c:432 console_thread() We've become an orphan, no more console for us. => According to the FAQ, running Freeswitch without a console requires using the "-nc" switch. Should I edit the rc.d script further to use this, or is the issue elsewhere? Thank you. From codecomplete at free.fr Wed Mar 25 04:56:55 2009 From: codecomplete at free.fr (Gilles) Date: Wed, 25 Mar 2009 12:56:55 +0100 Subject: [Freeswitch-users] Starting Freeswitch at boot-time with rc.d script? Message-ID: <7.0.1.0.2.20090325125455.02842df0@free.fr> Add-on: The script is almost fine now, but I don't know whether the script really requires a config file, and if yes, where it can be found (the XML file doesn't seem to be it): FREESWITCH_PARAMS="-nc" FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitch #BAD!!! FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml #. $FREESWITCH_CONFIG Thank you. From pablosaro at gmail.com Wed Mar 25 05:20:19 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Wed, 25 Mar 2009 09:20:19 -0300 Subject: [Freeswitch-users] Console Window In-Reply-To: References: <488035.39134.qm@web53604.mail.re2.yahoo.com> Message-ID: <247f8100903250520v1fb568bbh99d47c4bd88e0e40@mail.gmail.com> Also guys, I recommend to use a script to handle FS instead of directly executing the binary. FS provides some examples. On 3/24/09, Even Andr? Fiskvik wrote: > Please see documentation at: > http://wiki.freeswitch.org/wiki/Fs_cli > and > http://wiki.freeswitch.org/wiki/Mod_commands > > > Best regards, > Even Andr? > > On 24. mars. 2009, at 15.23, Will Smith wrote: > >> Thank you all for your help. >> I use : /usr/local/freeswitch/bin/fs_cli to open a FS instance. But >> then cannot use "shutdown" command. Then I used 'fsctl shutdown >> asap' (directed by Szymon). It works perfectly. >> >> Again, thank you. Have a great day you all. >> >> Will >> >> --- On Tue, 3/24/09, Leon de Rooij wrote: >> From: Leon de Rooij >> Subject: Re: [Freeswitch-users] Console Window >> To: freeswitch-users at lists.freeswitch.org >> Date: Tuesday, March 24, 2009, 6:47 AM >> >> Just make sure you have mod_event socket loaded in conf/ >> autoload_modules/modules.conf.xml : >> >> >> >> And have it configured in conf/autoload_modules/event_socket.conf.xml >> >> Then you can use bin/fs_cli to connect to running FS instance. >> >> I don't think you can reconnect to a process of which you have >> disconnected the terminal (without using screen).. >> >> regards, >> >> Leon >> >> On Mar 24, 2009, at 2:35 PM, Will Smith wrote: >> >>> No, I started FS with this: >>> /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? >>> >>> thank you for help >>> >>> --- On Tue, 3/24/09, Saeed Ahmed wrote: >>> From: Saeed Ahmed >>> Subject: Re: [Freeswitch-users] Console Window >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Tuesday, March 24, 2009, 6:25 AM >>> >>> Did you started FS with ?nc option? >>> with this option you can connect to FS using ./fs_cli >>> OR >>> use screen! >>> >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Will Smith >>> Sent: Tuesday, March 24, 2009 1:52 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: [Freeswitch-users] Console Window >>> >>> >>> Hi, >>> >>> I closed the console window that ran FS without shutting down FS. I >>> had to restart the server to get back to the console window. I just >>> wonder if there is a way to bring that window up without restart >>> server. In the file /log/freeswitch.pid, I found a number, is that >>> a seesion id? >>> >>> >>> >>> Thank you. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- Sent from Gmail for mobile | mobile.google.com From steve.d.ward at gmail.com Wed Mar 25 05:44:41 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Wed, 25 Mar 2009 08:44:41 -0400 Subject: [Freeswitch-users] conference calling ideas Message-ID: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> If any can share some ideas, I'm looking at making conference calls simple for the end-users of my FS system. Here are some issues I'm kicking around: 1. To create/join conferences - do you make a pre-defined list of extensions, each of which would join the caller to a particular conference room, or do you perhaps define one extension that will join the caller to a dynamically-named conference room (perhaps after making a choice in an IVR)? 2. How do you implement the idea of a conference leader - having a user who must call into a conference first before anybody else can join? Thanks for any ideas. - SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/1c716cfa/attachment.html From solko at gcdf.pl Wed Mar 25 06:02:17 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 25 Mar 2009 14:02:17 +0100 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> Message-ID: <49CA2B59.6070708@gcdf.pl> Steven Ward pisze: > If any can share some ideas, I'm looking at making conference > calls simple for the end-users of my FS system. > > Here are some issues I'm kicking around: > > 1. To create/join conferences - do you make a pre-defined list of > extensions, each of which would join the caller to a particular > conference room, or do you perhaps define one extension that will join > the caller to a dynamically-named conference room (perhaps after making > a choice in an IVR)? > > 2. How do you implement the idea of a conference leader - having a > user who must call into a conference first before anybody else can join? > > Thanks for any ideas. > Can you give more info what you plan to do? If you don't care for conferences name and they can be numbers then just make one extension and then use that extension or part of it to name conference. If you want to make conference leader then make 2 separate extensions, in second extension which is used for not leaders use that conference command. 'conference xxx bgdial '.In current SVN, bgdial does not create new conference it must exists to make that command. This can be make just by dial plan. If you plan to control FS from any language then you have much more abilities. > - SW > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steve.d.ward at gmail.com Wed Mar 25 06:37:45 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Wed, 25 Mar 2009 09:37:45 -0400 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <49CA2B59.6070708@gcdf.pl> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> <49CA2B59.6070708@gcdf.pl> Message-ID: <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> Szymon, I want to provide a service wherein a user can reserve a teleconference room for a partiuclar time and control who can join the conference call (only those invited). I want to support several of such conference calls at any given time. I want callers who were not invited to a conference call, but who try to call in, to be notified they can't join. Also, it should be simple enough that the user just has to dial a number and be able to know what to do from there to join the conference. If the conference leader is late to the conference call and others try to call in (before the conference is created), the callers should be notified the conf leader hasn't started the conference yet. Perhaps they can be told to try later, or they can be put on hold. These are the kinds of things I need to think about implementing, and I do not know yet what other features would be useful. Thanks. On Wed, Mar 25, 2009 at 9:02 AM, Szymon Olko wrote: > Steven Ward pisze: > > If any can share some ideas, I'm looking at making conference > > calls simple for the end-users of my FS system. > > > > Here are some issues I'm kicking around: > > > > 1. To create/join conferences - do you make a pre-defined list of > > extensions, each of which would join the caller to a particular > > conference room, or do you perhaps define one extension that will join > > the caller to a dynamically-named conference room (perhaps after making > > a choice in an IVR)? > > > > 2. How do you implement the idea of a conference leader - having a > > user who must call into a conference first before anybody else can join? > > > > Thanks for any ideas. > > > Can you give more info what you plan to do? > > If you don't care for conferences name and they can be numbers then just > make one extension and then use that extension or part of > it to name conference. > > If you want to make conference leader then make 2 separate extensions, in > second extension which is used for not leaders use that > conference command. 'conference xxx bgdial '.In current SVN, > bgdial does not create new conference it must exists to > make that command. > > This can be make just by dial plan. If you plan to control FS from any > language then you have much more abilities. > > > - SW > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/f779404d/attachment.html From solko at gcdf.pl Wed Mar 25 07:02:32 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 25 Mar 2009 15:02:32 +0100 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> <49CA2B59.6070708@gcdf.pl> <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> Message-ID: <49CA3978.7020704@gcdf.pl> Steven Ward pisze: > Szymon, I want to provide a service wherein a user can reserve a > teleconference room for a partiuclar time and control who can join the > conference call (only those invited). I want to support several of > such conference calls at any given time. > > I want callers who were not invited to a conference call, but who try to > call in, to be notified they can't join. > Reserving conference means some storage of it. I think you have write some application to control it. You can use store that in dialplan but it will be hard to make such dialplan. There is pin code for conference so you can use it. > Also, it should be simple enough that the user just has to dial a number > and be able to know what to do from there to join the conference. > > If the conference leader is late to the conference call and others try > to call in (before the conference is created), the callers should be > notified the conf leader hasn't started the conference yet. Perhaps > they can be told to try later, or they can be put on hold. > > These are the kinds of things I need to think about implementing, and I > do not know yet what other features would be useful. Thanks. I would make it like that. If you need to limit total number of conferences then you have to manage time of reservation. If you don't need that then make one extension in which you do the following. - get pincode from user. - get admin_pincode from user. - look in db for next available conference number - store conference number and pincode in database. - return that number to user. Make second extension where you do folowing: - get conference number from user - get pincode from user - check in db if conference and pincode are the same as stored in db. - check in database if admin is in conference. Make third extension: - get conference number from user - get admin pincode from user - check for matching in db - transfer to conference - store in db that admin is online With this way users must get conference name and pincode to access. You can change it and in first extension add ability for user to add others members with their personal number or something like that. Szymon Olko > On Wed, Mar 25, 2009 at 9:02 AM, Szymon Olko wrote: > > Steven Ward pisze: > > If any can share some ideas, I'm looking at making conference > > calls simple for the end-users of my FS system. > > > > Here are some issues I'm kicking around: > > > > 1. To create/join conferences - do you make a pre-defined list of > > extensions, each of which would join the caller to a particular > > conference room, or do you perhaps define one extension that will join > > the caller to a dynamically-named conference room (perhaps after > making > > a choice in an IVR)? > > > > 2. How do you implement the idea of a conference leader - having a > > user who must call into a conference first before anybody else can > join? > > > > Thanks for any ideas. > > > Can you give more info what you plan to do? > > If you don't care for conferences name and they can be numbers then > just make one extension and then use that extension or part of > it to name conference. > > If you want to make conference leader then make 2 separate > extensions, in second extension which is used for not leaders use that > conference command. 'conference xxx bgdial '.In current > SVN, bgdial does not create new conference it must exists to > make that command. > > This can be make just by dial plan. If you plan to control FS from > any language then you have much more abilities. > > > - SW > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From congxin.zhao at gmail.com Wed Mar 25 03:52:50 2009 From: congxin.zhao at gmail.com (congxin zhao) Date: Wed, 25 Mar 2009 18:52:50 +0800 Subject: [Freeswitch-users] help about the rtp port. In-Reply-To: References: Message-ID: Hi, I meet a issue about the rtp port. The media port value in the sdp of SIP 200 OK which freeswitch internal sends to ua is always 0, so the media forward is always wrong. What could cause this problem possibly? No matter I set or The problem always showes. Does the following rtp-ip and ext-rtp-ip set correct? In external.xml, I set both rtp-ip and ext-rtp-ip as $${external_rtp_ip}, is that correct? I am not clear about this two variables in the external.xml and internal.xml external.xml: external.xml: internal.xml: internal.xml: Thanks, -Congxin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/fbbea456/attachment.html From dujinfang at gmail.com Wed Mar 25 08:25:12 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 25 Mar 2009 23:25:12 +0800 Subject: [Freeswitch-users] Starting Freeswitch at boot-time with rc.d script? In-Reply-To: <7.0.1.0.2.20090325125455.02842df0@free.fr> References: <7.0.1.0.2.20090325125455.02842df0@free.fr> Message-ID: <6854FBCC-B8D9-4947-8662-9E1A30680B57@gmail.com> I don't think you need FREESWITCH_CONFIG. It will find all configuration files on default place, say /usr/local/freeswitch/conf If you store config files in other place, the command line should like this /usr/local/freeswitch/bin/freeswitch -conf /tmp/conf -db /tmp/db -log / tmp/log On Mar 25, 2009, at 7:56 PM, Gilles wrote: > Add-on: The script is almost fine now, but I don't know whether the > script really requires a config file, and if yes, where it can be > found (the XML file doesn't seem to be it): > > FREESWITCH_PARAMS="-nc" > FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitchI > > #BAD!!! FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml > #. $FREESWITCH_CONFIG > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mattdfong at gmail.com Wed Mar 25 08:26:30 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 25 Mar 2009 22:26:30 +0700 Subject: [Freeswitch-users] Cron-like execution in FS Message-ID: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com> I'm wondering if there's any features that allow the cron-like execution of code inside of Freeswitch, preferably with lua--or if I am stuck using the api interface and running the cron outside of freeswitch. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/45a46f45/attachment.html From pablosaro at gmail.com Wed Mar 25 08:43:38 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Wed, 25 Mar 2009 12:43:38 -0300 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> References: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> Message-ID: <247f8100903250843l9fa97a8oa96adae8c05d6f01@mail.gmail.com> First of all, thanks for your answers. You guys are awesome and FS rocks. Don't take it the wrong way... My custom is to install production systems using stable versions and apply the corresponding security patches as soon as they are released. But I don't change software versions until I really need to do it (for example when I need a new functionality). I've installed production systems using FS SVN trunk, but it changes every day. So, to keep it up to date I have to re-compile it every week at least. I was wondering if there is a version you can affirm has no major bugs. If not, there is no problem with that... Just wondering... Pablo On Mon, Mar 23, 2009 at 8:32 PM, Brian West wrote: > Well put! > > /b > > On Mar 23, 2009, at 6:31 PM, Ken Rice wrote: > >> >> Until we get more people testing and actually reporting bugs on RCs >> I doubt >> this well ever change... > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Mark.Tabron at rnid-typetalk.org.uk Wed Mar 25 08:49:59 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Wed, 25 Mar 2009 15:49:59 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765381@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765382@tt-mail.RNID.TYPETALK.LOCAL> <200903201929.05829.stkn@freeswitch.org> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765389@tt-mail.RNID.TYPETALK.LOCAL> Thanks for the additional info. Found a problem with ozmod_libri.so missing from the mod directory after installation (fixed by manually copying file over) and I can now happily confirm I can make calls both in and out! Thanks for all the help on this - really appreciated. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stefan Knoblich Sent: 20 March 2009 18:29 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Am Friday 20 March 2009 schrieb Mark Tabron: > Installed libpri but I'm stuck on what entries to put in > openzap.conf.xml, here's how I have the span setup at the moment: > > > > > > > > > > > Node and Switch type aren't documented for libpri from what I can tell - > I know the former is either CPE or NET, though, I'm unsure what other > values can be used for switch type. > The value for switch is invalid, it's going to fall back to dms100 with that one set. Valid settings are: ni1, ni2, dms100, euroisdn, lucent5e, att4ess, gr303eoc and gr303tmc and "euroisdn" is the one you'll want for a E1 line. Another setting you may need is: stkn -- ------------------------------------------------------------------------ ------- Stefan Knoblich | axsentis GmbH | Web: http://www.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | http://oss.axsentis.de/ Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------ ------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- From brian at freeswitch.org Wed Mar 25 09:35:54 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 11:35:54 -0500 Subject: [Freeswitch-users] help about the rtp port. In-Reply-To: References: Message-ID: <58A60265-2CBF-409A-8483-7371F82AD4B6@freeswitch.org> Zero for the port means something please provide a complete sip trace. /b On Mar 25, 2009, at 5:52 AM, congxin zhao wrote: > Hi, > > I meet a issue about the rtp port. The media port value in the sdp > of SIP 200 OK which freeswitch internal sends to ua is always 0, so > the media forward is always wrong. What could cause this problem > possibly? > > No matter I set > > or > > The problem always showes. > > Does the following rtp-ip and ext-rtp-ip set correct? In > external.xml, I set both rtp-ip and ext-rtp-ip as $$ > {external_rtp_ip}, is that correct? I am not clear about this two > variables in the external.xml and internal.xml > > external.xml: > external.xml: > > internal.xml: > internal.xml: > > > Thanks, > -Congxin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/ff5f22a3/attachment.html From msc at freeswitch.org Wed Mar 25 09:40:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 09:40:02 -0700 Subject: [Freeswitch-users] Cron-like execution in FS In-Reply-To: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com> References: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com> Message-ID: <87f2f3b90903250940s4bf14381m8bb78d69f2f922fe@mail.gmail.com> 2009/3/25 Matthew Fong : > I'm wondering if there's any features that allow the cron-like execution of > code inside of Freeswitch,?preferably?with lua--or if I am stuck using the > api interface and running the cron outside of freeswitch. > --matt I guess the most important question you can answer is this: why do you need the cron-like feature to be inside FreeSWITCH? -MC From msc at freeswitch.org Wed Mar 25 10:03:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 10:03:49 -0700 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <247f8100903250843l9fa97a8oa96adae8c05d6f01@mail.gmail.com> References: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> <247f8100903250843l9fa97a8oa96adae8c05d6f01@mail.gmail.com> Message-ID: <87f2f3b90903251003i109fc3c7k3ad129308d6343de@mail.gmail.com> On Wed, Mar 25, 2009 at 8:43 AM, Pablo Hernan Saro wrote: > First of all, thanks for your answers. You guys are awesome and FS > rocks. Don't take it the wrong way... No offense taken. > My custom is to install production systems using stable versions and > apply the corresponding security patches as soon as they are released. > But I don't change software versions until I really need to do it (for > example when I need a new functionality). Nothing wrong with this custom. > I've installed production systems using FS SVN trunk, but it changes > every day. So, to keep it up to date I have to re-compile it every > week at least. Understood. Production systems have their own issues. One custom that is common is to have a production system and a "sandbox" system. You could have your sandbox system to a "make current" every night and you could run simple tests via cron to see if anything important to you is somehow broken. If all is well then you know that you can upgrade your production system. > I was wondering if there is a version you can affirm has no major > bugs. If not, there is no problem with that... Just wondering... This is a tough one. It may be a minor bug to me but it could be a major bug to you. In other words this is almost an impossible question to answer. This much I can say: the FS devs make lots of changes each day but they don't leave the SVN trunk in an unstable state at any point, especially at the end of the day. (GMT -6 time zone.) The bottom line is still the same: FreeSWITCH is one of those rare software projects where the SVN trunk is almost always more stable and less buggy than the most recent "stable" release. My recommendation to you is to find the update schedule that works best for you, be it daily, weekly or some other period. -MC From chris at fowler.cc Wed Mar 25 11:19:09 2009 From: chris at fowler.cc (Chris Fowler) Date: Wed, 25 Mar 2009 11:19:09 -0700 Subject: [Freeswitch-users] DTMF Missing Digits Message-ID: <1238005149.12393.1307283305@webmail.messagingengine.com> Any thoughts on why FS saw all digits "1029" but only reports '029'? 2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 play_and_collect() digits '029' Config: Trace: 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1301 do_2833() Send start packet for [1] ts=1129880426 dur=160/160/2000 seq=2804 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=320/320/2000 seq=2805 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=480/480/2000 seq=2806 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=640/640/2000 seq=2807 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=800/800/2000 seq=2808 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=960/960/2000 seq=2809 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1120/1120/2000 seq=2810 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1280/1280/2000 seq=2811 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1440/1440/2000 seq=2812 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1600/1600/2000 seq=2813 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1760/1760/2000 seq=2814 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1920/1920/2000 seq=2815 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [1] ts=1129880426 dur=2080/2080/2000 seq=2816 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [1] ts=1129880426 dur=2080/2080/2000 seq=2817 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [1] ts=1129880426 dur=2080/2080/2000 seq=2818 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 0:2160 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1301 do_2833() Send start packet for [0] ts=1129884426 dur=160/160/2160 seq=2819 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=320/320/2160 seq=2820 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=480/480/2160 seq=2821 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=640/640/2160 seq=2822 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=800/800/2160 seq=2823 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=960/960/2160 seq=2824 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1120/1120/2160 seq=2825 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1280/1280/2160 seq=2826 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1440/1440/2160 seq=2827 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1600/1600/2160 seq=2828 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1760/1760/2160 seq=2829 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1920/1920/2160 seq=2830 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=2080/2080/2160 seq=2831 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [0] ts=1129884426 dur=2240/2240/2160 seq=2832 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [0] ts=1129884426 dur=2240/2240/2160 seq=2833 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [0] ts=1129884426 dur=2240/2240/2160 seq=2834 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 2:2000 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1301 do_2833() Send start packet for [2] ts=1129887626 dur=160/160/2000 seq=2835 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=320/320/2000 seq=2836 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=480/480/2000 seq=2837 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=640/640/2000 seq=2838 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=800/800/2000 seq=2839 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=960/960/2000 seq=2840 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1120/1120/2000 seq=2841 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1280/1280/2000 seq=2842 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1440/1440/2000 seq=2843 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1600/1600/2000 seq=2844 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1760/1760/2000 seq=2845 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 0:2080 2009-03-25 10:48:42 [DEBUG] switch_ivr_play_say.c:1280 switch_ivr_play_file() done playing file 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1920/1920/2000 seq=2846 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [2] ts=1129887626 dur=2080/2080/2000 seq=2847 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [2] ts=1129887626 dur=2080/2080/2000 seq=2848 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [2] ts=1129887626 dur=2080/2080/2000 seq=2849 2009-03-25 10:48:42 [DEBUG] switch_ivr_menu.c:319 play_and_collect() waiting for 3/4 digits t/o 1500 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 9:2000 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1301 do_2833() Send start packet for [9] ts=1129890826 dur=160/160/2000 seq=2850 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=320/320/2000 seq=2851 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=480/480/2000 seq=2852 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=640/640/2000 seq=2853 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=800/800/2000 seq=2854 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=960/960/2000 seq=2855 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1120/1120/2000 seq=2856 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1280/1280/2000 seq=2857 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1440/1440/2000 seq=2858 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1600/1600/2000 seq=2859 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1760/1760/2000 seq=2860 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1920/1920/2000 seq=2861 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [9] ts=1129890826 dur=2080/2080/2000 seq=2862 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [9] ts=1129890826 dur=2080/2080/2000 seq=2863 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [9] ts=1129890826 dur=2080/2080/2000 seq=2864 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 2:2080 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 9:2080 2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 play_and_collect() digits '029' From brian at freeswitch.org Wed Mar 25 11:27:18 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 13:27:18 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238005149.12393.1307283305@webmail.messagingengine.com> References: <1238005149.12393.1307283305@webmail.messagingengine.com> Message-ID: <7BE6FEF3-E132-4DA2-939B-85D6590DA757@freeswitch.org> First off what SVN rev? Remember when reporting issues try to include all the information you can! /b On Mar 25, 2009, at 1:19 PM, Chris Fowler wrote: > Any thoughts on why FS saw all digits "1029" but only reports '029'? > 2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 > play_and_collect() > digits '029' From chris at fowler.cc Wed Mar 25 11:49:45 2009 From: chris at fowler.cc (Chris Fowler) Date: Wed, 25 Mar 2009 11:49:45 -0700 Subject: [Freeswitch-users] DTMF Missing Digits Message-ID: <1238006985.18397.1307291893@webmail.messagingengine.com> >> First off what SVN rev? Remember when reporting issues try to include all the information you can! Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647) From brian at freeswitch.org Wed Mar 25 11:53:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 13:53:21 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238006985.18397.1307291893@webmail.messagingengine.com> References: <1238006985.18397.1307291893@webmail.messagingengine.com> Message-ID: <4604EEB6-4F7D-427A-9C19-15DC3BF3CB8B@freeswitch.org> Please review this link http://wiki.freeswitch.org/wiki/Reporting_Bugs The rules are try to reproduce this on SVN Trunk... I am pretty sure we fixed this one already. /b On Mar 25, 2009, at 1:49 PM, Chris Fowler wrote: > Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647) From pablosaro at gmail.com Wed Mar 25 11:55:06 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Wed, 25 Mar 2009 15:55:06 -0300 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <87f2f3b90903251003i109fc3c7k3ad129308d6343de@mail.gmail.com> References: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> <247f8100903250843l9fa97a8oa96adae8c05d6f01@mail.gmail.com> <87f2f3b90903251003i109fc3c7k3ad129308d6343de@mail.gmail.com> Message-ID: <247f8100903251155k41472db1ja353ffe6676cd5e7@mail.gmail.com> Hi Michael, All I have to say is thank you very much. Thanks for this explanation, that's what I was looking for. So, I will figure out which is the most convenient update schedule for me and I will build a sandbox for this exclusive purpose. Not only FS rocks, but the community too. Kudos Pablo On Wed, Mar 25, 2009 at 2:03 PM, Michael Collins wrote: > On Wed, Mar 25, 2009 at 8:43 AM, Pablo Hernan Saro wrote: >> First of all, thanks for your answers. You guys are awesome and FS >> rocks. Don't take it the wrong way... > > No offense taken. > >> My custom is to install production systems using stable versions and >> apply the corresponding security patches as soon as they are released. >> But I don't change software versions until I really need to do it (for >> example when I need a new functionality). > > Nothing wrong with this custom. > >> I've installed production systems using FS SVN trunk, but it changes >> every day. So, to keep it up to date I have to re-compile it every >> week at least. > > Understood. Production systems have their own issues. One custom that > is common is to have a production system and a "sandbox" system. You > could have your sandbox system to a "make current" every night and you > could run simple tests via cron to see if anything important to you is > somehow broken. If all is well then you know that you can upgrade your > production system. > >> I was wondering if there is a version you can affirm has no major >> bugs. If not, there is no problem with that... Just wondering... > > This is a tough one. It may be a minor bug to me but it could be a > major bug to you. In other words this is almost an impossible question > to answer. This much I can say: the FS devs make lots of changes each > day but they don't leave the SVN trunk in an unstable state at any > point, especially at the end of the day. (GMT -6 time zone.) > > The bottom line is still the same: FreeSWITCH is one of those rare > software projects where the SVN trunk is almost always more stable and > less buggy than the most recent "stable" release. My recommendation to > you is to find the update schedule that works best for you, be it > daily, weekly or some other period. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jforman at wcgltd.com Wed Mar 25 12:01:01 2009 From: jforman at wcgltd.com (Josh Forman) Date: Wed, 25 Mar 2009 15:01:01 -0400 Subject: [Freeswitch-users] Question about FAQ question Message-ID: I'm running Freeswitch on Ubuntu 64bit Intrepid and on a svn rev 12722, freeswitch would install and run fine but as soon as calls started coming in it would have a segmentation fault. This is the second svn snapshot I've had this happen on. In the Freeswitch FAQs there is a question concerning segmentation fault on ubuntu 64bit except they say it occurs on start. Their solution is to recompile libedit which I plan to try regardless, but I just wanted to know if the scenario they refer to is the same as what I'm experiencing or was there some other problem where freeswitch would segfault while initially loading. Until then I'm still running rev 12289 on the system that is having issues. Thanks -Josh From brian at freeswitch.org Wed Mar 25 12:07:24 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 14:07:24 -0500 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: References: Message-ID: On Mar 25, 2009, at 2:01 PM, Josh Forman wrote: > I'm running Freeswitch on Ubuntu 64bit Intrepid and on a svn rev > 12722, freeswitch would install and run fine but as soon as calls > started coming in it would have a segmentation fault. This is the > second svn snapshot I've had this happen on. Please review http://wiki.freeswitch.org/wiki/Reporting_Bugs Collect the info and report it on jira. If it still happens on SVN trunk as of NOW then please collect the back trace as per the reporting bugs guide and we'll investigate the issue. Also be aware if you aren't doing a "make current", you could have build skew which could be the whole problem all along. Incorrectly updating your system will result in all kinds of strange behaviors. > In the Freeswitch FAQs there is a question concerning segmentation > fault on ubuntu 64bit except they say it occurs on start. Their > solution is to recompile libedit which I plan to try regardless, but I > just wanted to know if the scenario they refer to is the same as what > I'm experiencing or was there some other problem where freeswitch > would segfault while initially loading. I don't think this is the problem you're having. > > Until then I'm still running rev 12289 on the system that is having > issues. This is a rather old REV, How about you report your back trace to jira as per the instructions above. > > Thanks > -Josh > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Mar 25 12:42:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 14:42:50 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238006985.18397.1307291893@webmail.messagingengine.com> References: <1238006985.18397.1307291893@webmail.messagingengine.com> Message-ID: <10BEDCFA-A9C8-4157-A5C0-45CCA42199AB@freeswitch.org> btw you'll have to reinstall your phrase macros .... make vm-sync I think should do it if it doesn't let me know... we removed the 250ms sleeps and that was the problem which we fixed. /b On Mar 25, 2009, at 1:49 PM, Chris Fowler wrote: >>> First off what SVN rev? Remember when reporting issues try to >>> include all the information you can! > > Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647) Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com From mszlazak at aol.com Wed Mar 25 12:50:58 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 25 Mar 2009 15:50:58 -0400 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: References: Message-ID: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> I'd like to do something like "Make current" but for Windows because I'm finding bugs to report.? One was on how the dialplan's extensions are being parsed. Extensions in some cases like when doing originates with sched_api, loose their last character and I have to add a white space after to solve the problem. Another issue is that dialing one extension landed me into another totally different extension. I had to comment it out to target the right extension. Maybe the reporting bugs wiki needs updating for Windows users (experienced and inexperienced). Thanks. Mark. ? -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Wed, 25 Mar 2009 12:07 pm Subject: Re: [Freeswitch-users] Question about FAQ question On Mar 25, 2009, at 2:01 PM, Josh Forman wrote: > I'm running Freeswitch on Ubuntu 64bit Intrepid and on a svn rev > 12722, freeswitch would install and run fine but as soon as calls > started coming in it would have a segmentation fault. This is the > second svn snapshot I've had this happen on. Please review http://wiki.freeswitch.org/wiki/Reporting_Bugs Collect the info and report it on jira. If it still happens on SVN trunk as of NOW then please collect the back trace as per the reporting bugs guide and we'll investigate the issue. Also be aware if you aren't doing a "make current", you could have build skew which could be the whole problem all along. Incorrectly updating your system will result in all kinds of strange behaviors. > In the Freeswitch FAQs there is a question concerning segmentation > fault on ubuntu 64bit except they say it occurs on start. Their > solution is to recompile libedit which I plan to try regardless, but I > just wanted to know if the scenario they refer to is the same as what > I'm experiencing or was there some other problem where freeswitch > would segfault while initially loading. I don't think this is the problem you're having. > > Until then I'm still running rev 12289 on the system that is having > issues. This is a rather old REV, How about you report your back trace to jira as per the instructions above. > > Thanks > -Josh > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/46dc4a0c/attachment.html From brian at freeswitch.org Wed Mar 25 12:55:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 14:55:14 -0500 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> References: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> Message-ID: <08EB85E2-4B2F-4A2D-99DB-29F0F753AB4A@freeswitch.org> If you svn up clean the solution and rebuild its the same thing.... Can you provide me a test case of this happening? /b On Mar 25, 2009, at 2:50 PM, mszlazak at aol.com wrote: > I'd like to do something like "Make current" but for Windows because > I'm finding bugs to report. > > One was on how the dialplan's extensions are being parsed. > Extensions in some cases like when doing originates with sched_api, > loose their last character and I have to add a white space after to > solve the problem. Another issue is that dialing one extension > landed me into another totally different extension. I had to comment > it out to target the right extension. > Maybe the reporting bugs wiki needs updating for Windows users > (experienced and inexperienced). > > Thanks. Mark. > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/df04f9b2/attachment-0001.html From msc at freeswitch.org Wed Mar 25 12:59:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 12:59:31 -0700 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> References: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> Message-ID: <87f2f3b90903251259w45ae04c0y5f81bfdebb131c54@mail.gmail.com> > Maybe the reporting bugs wiki needs updating for Windows users (experienced > and inexperienced). Quite possibly. The instructions are not explicit. I will add something for the Windows users that's a bit more specific. -MC From mszlazak at aol.com Wed Mar 25 13:37:50 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 25 Mar 2009 16:37:50 -0400 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: <87f2f3b90903251259w45ae04c0y5f81bfdebb131c54@mail.gmail.com> References: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> <87f2f3b90903251259w45ae04c0y5f81bfdebb131c54@mail.gmail.com> Message-ID: <8CB7BA0B282A3A3-424-7D@WEBMAIL-MB17.sysops.aol.com> Thanks MC, maybe a link to that "TortoiseSVN" would help for some in the Windows crowd. TortoiseSVN has a bunch of stuff in it but to make it simple, especially for doing updates to report bugs, then mentioning if doing just an "SVN update" will work before rebuild. Also, what to do if one gets an errror using SVN Update. I got one about not being able to open a file. So, I didn't know if any of the rest of the SVN update succeeded. I guess it wasn't a "clean" update. I didn't bother rebuilding afterwards since I also didn't know if it would work. There was to much to go through in TortoiseSVN documentation for the time I had so I didn't report the errors and left things for later when I would just download and install a newer version of FS. It sounds lazy but as an inexperienced user it's enough discouragment to let these things go. Anyway, just a bit more instructions might get more bugs reported. Thanks again. Mark. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 25 Mar 2009 12:59 pm Subject: Re: [Freeswitch-users] Question about FAQ question > Maybe the reporting bugs wiki needs updating for Windows users (experienced > and inexperienced). Quite possibly. The instructions are not explicit. I will add something for the Windows users that's a bit more specific. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/61610745/attachment.html From msc at freeswitch.org Wed Mar 25 13:48:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 13:48:51 -0700 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: <8CB7BA0B282A3A3-424-7D@WEBMAIL-MB17.sysops.aol.com> References: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> <87f2f3b90903251259w45ae04c0y5f81bfdebb131c54@mail.gmail.com> <8CB7BA0B282A3A3-424-7D@WEBMAIL-MB17.sysops.aol.com> Message-ID: <87f2f3b90903251348v1f47b8eau34fcb4d8c8d2759f@mail.gmail.com> 2009/3/25 : > Thanks MC, maybe a link to that "TortoiseSVN" would help for some in the > Windows crowd. Done! > > TortoiseSVN has a bunch of stuff in it but to make it simple, especially for > doing updates to report bugs, then mentioning if doing just an "SVN update" > will work before rebuild. Also, what to do if one gets an errror using SVN > Update. > I got one about not being able to open a file. So, I didn't know if any of > the rest of the SVN update succeeded. I guess it wasn't a "clean" update. > I didn't bother rebuilding afterwards since I also didn't know if it would > work. There was to much to go through in TortoiseSVN documentation for the > time I had so I didn't report the errors and left things for later when I > would just download and install a newer version of FS. > It sounds lazy but as an inexperienced user it's enough discouragment to let > these things go. > Anyway, just a bit more instructions might get more bugs reported. Understood. We can't anticipate every possible error and give documentation otherwise the Reporting Bugs page would be a lot larger than it is. In cases like this I recommend the panacea for all ills: Google. :) -MC From msc at freeswitch.org Wed Mar 25 13:53:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 13:53:02 -0700 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <49CA3978.7020704@gcdf.pl> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> <49CA2B59.6070708@gcdf.pl> <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> <49CA3978.7020704@gcdf.pl> Message-ID: <87f2f3b90903251353g1fe1e1d7i874e2593ed2c593c@mail.gmail.com> Hey, if you guys get this all figured out, tested, and working then please be sure to put it on the wiki. You could create a whole new page and then link to/from the mod_conference page. -MC On Wed, Mar 25, 2009 at 7:02 AM, Szymon Olko wrote: > Steven Ward pisze: >> Szymon, I want to provide a service wherein a user can reserve a >> teleconference room for a partiuclar time and control who can join the >> conference call (only those invited). ?I want to support several of >> such conference calls at any given time. >> >> I want callers who were not invited to a conference call, but who try to >> call in, to be notified they can't join. >> > Reserving conference means some storage of it. I think you have write some application to control it. > You can use store that in dialplan but it will be hard to make such dialplan. > There is pin code for conference so you can use it. > >> Also, it should be simple enough that the user just has to dial a number >> and be able to know what to do from there to join the conference. >> >> If the conference leader is late to the conference call and others try >> to call in (before the conference is created), the callers should be >> notified the conf leader hasn't started the conference yet. ?Perhaps >> they can be told to try later, or they can be put on hold. >> >> These are the kinds of things I need to think about implementing, and I >> do not know yet what other features would be useful. ?Thanks. > > I would make it like that. If you need to limit total number of conferences then you have to manage time of reservation. If you > don't need that then make one extension in which you do the following. > - get pincode from user. > - get admin_pincode from user. > - look in db for next available conference number > - store conference number and pincode in database. > - return that number to user. > > Make second extension where you do folowing: > - get conference number from user > - get pincode from user > - check in db if conference and pincode are the same as stored in db. > - check in database if admin is in conference. > > > Make third extension: > - get conference number from user > - get admin pincode from user > - check for matching in db > - transfer to conference > - store in db that admin is online > > > With this way users must get conference name and pincode to access. You can change it and in first extension add ability for user > to add others members with their personal number or something like that. > > Szymon Olko > >> On Wed, Mar 25, 2009 at 9:02 AM, Szymon Olko wrote: >> >> ? ? Steven Ward pisze: >> ? ? > If any can share some ideas, I'm looking at making conference >> ? ? > calls simple for the end-users of my FS system. >> ? ? > >> ? ? > Here are some issues I'm kicking around: >> ? ? > >> ? ? > 1. ?To create/join conferences - do you make a pre-defined list of >> ? ? > extensions, each of which would join the caller to a particular >> ? ? > conference room, or do you perhaps define one extension that will join >> ? ? > the caller to a dynamically-named conference room (perhaps after >> ? ? making >> ? ? > a choice in an IVR)? >> ? ? > >> ? ? > 2. ?How do you implement the idea of a conference leader - having a >> ? ? > user who must call into a conference first before anybody else can >> ? ? join? >> ? ? > >> ? ? > Thanks for any ideas. >> ? ? > >> ? ? Can you give more info what you plan to do? >> >> ? ? If you don't care for conferences name and they can be numbers then >> ? ? just make one extension and then use that extension or part of >> ? ? it to name conference. >> >> ? ? If you want to make conference leader then make 2 separate >> ? ? extensions, in second extension which is used for not leaders use that >> ? ? conference command. 'conference xxx bgdial '.In current >> ? ? SVN, bgdial does not create new conference it must exists to >> ? ? make that command. >> >> ? ? This can be make just by dial plan. If you plan to control FS from >> ? ? any language then you have much more abilities. >> >> ? ? > - SW >> ? ? > >> ? ? > >> ? ? > >> ? ? ------------------------------------------------------------------------ >> ? ? > >> ? ? > _______________________________________________ >> ? ? > Freeswitch-users mailing list >> ? ? > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steve.d.ward at gmail.com Wed Mar 25 14:03:23 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Wed, 25 Mar 2009 17:03:23 -0400 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <87f2f3b90903251353g1fe1e1d7i874e2593ed2c593c@mail.gmail.com> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> <49CA2B59.6070708@gcdf.pl> <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> <49CA3978.7020704@gcdf.pl> <87f2f3b90903251353g1fe1e1d7i874e2593ed2c593c@mail.gmail.com> Message-ID: <4ea6e8f20903251403k37ea8331y7087be3b59edc2e@mail.gmail.com> I need some time to work out my setup and explore some different options, but I'll be happy to get something together for the wiki on this as soon as I'm able. Thanks. On Wed, Mar 25, 2009 at 4:53 PM, Michael Collins wrote: > Hey, if you guys get this all figured out, tested, and working then > please be sure to put it on the wiki. You could create a whole new > page and then link to/from the mod_conference page. > -MC > > On Wed, Mar 25, 2009 at 7:02 AM, Szymon Olko wrote: > > Steven Ward pisze: > >> Szymon, I want to provide a service wherein a user can reserve a > >> teleconference room for a partiuclar time and control who can join the > >> conference call (only those invited). I want to support several of > >> such conference calls at any given time. > >> > >> I want callers who were not invited to a conference call, but who try to > >> call in, to be notified they can't join. > >> > > Reserving conference means some storage of it. I think you have write > some application to control it. > > You can use store that in dialplan but it will be hard to make such > dialplan. > > There is pin code for conference so you can use it. > > > >> Also, it should be simple enough that the user just has to dial a number > >> and be able to know what to do from there to join the conference. > >> > >> If the conference leader is late to the conference call and others try > >> to call in (before the conference is created), the callers should be > >> notified the conf leader hasn't started the conference yet. Perhaps > >> they can be told to try later, or they can be put on hold. > >> > >> These are the kinds of things I need to think about implementing, and I > >> do not know yet what other features would be useful. Thanks. > > > > I would make it like that. If you need to limit total number of > conferences then you have to manage time of reservation. If you > > don't need that then make one extension in which you do the following. > > - get pincode from user. > > - get admin_pincode from user. > > - look in db for next available conference number > > - store conference number and pincode in database. > > - return that number to user. > > > > Make second extension where you do folowing: > > - get conference number from user > > - get pincode from user > > - check in db if conference and pincode are the same as stored in db. > > - check in database if admin is in conference. > > > > > > Make third extension: > > - get conference number from user > > - get admin pincode from user > > - check for matching in db > > - transfer to conference > > - store in db that admin is online > > > > > > With this way users must get conference name and pincode to access. You > can change it and in first extension add ability for user > > to add others members with their personal number or something like that. > > > > Szymon Olko > > > >> On Wed, Mar 25, 2009 at 9:02 AM, Szymon Olko wrote: > >> > >> Steven Ward pisze: > >> > If any can share some ideas, I'm looking at making conference > >> > calls simple for the end-users of my FS system. > >> > > >> > Here are some issues I'm kicking around: > >> > > >> > 1. To create/join conferences - do you make a pre-defined list of > >> > extensions, each of which would join the caller to a particular > >> > conference room, or do you perhaps define one extension that will > join > >> > the caller to a dynamically-named conference room (perhaps after > >> making > >> > a choice in an IVR)? > >> > > >> > 2. How do you implement the idea of a conference leader - having > a > >> > user who must call into a conference first before anybody else can > >> join? > >> > > >> > Thanks for any ideas. > >> > > >> Can you give more info what you plan to do? > >> > >> If you don't care for conferences name and they can be numbers then > >> just make one extension and then use that extension or part of > >> it to name conference. > >> > >> If you want to make conference leader then make 2 separate > >> extensions, in second extension which is used for not leaders use > that > >> conference command. 'conference xxx bgdial '.In current > >> SVN, bgdial does not create new conference it must exists to > >> make that command. > >> > >> This can be make just by dial plan. If you plan to control FS from > >> any language then you have much more abilities. > >> > >> > - SW > >> > > >> > > >> > > >> > ------------------------------------------------------------------------ > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/b3b1940c/attachment-0001.html From emercado at rapidlink.com Wed Mar 25 13:01:17 2009 From: emercado at rapidlink.com (chevio) Date: Wed, 25 Mar 2009 13:01:17 -0700 (PDT) Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> Message-ID: <22709805.post@talk.nabble.com> How was this fixed ?, I am experiencing the same problem. Chevio Shelby Ramsey-2 wrote: > > Thanks for the help. That did the trick. > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Compile-Errors-...-tp22149989p22709805.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Wed Mar 25 14:39:21 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Mar 2009 17:39:21 -0400 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <22709805.post@talk.nabble.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> <22709805.post@talk.nabble.com> Message-ID: it was never really fixed as no one let me into their machine to troubleshoot. Mike On Mar 25, 2009, at 4:01 PM, chevio wrote: > > How was this fixed ?, I am experiencing the same problem. > > Chevio > > > Shelby Ramsey-2 wrote: >> >> Thanks for the help. That did the trick. >> >> SDR >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Compile-Errors-...-tp22149989p22709805.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From congxin.zhao at gmail.com Wed Mar 25 18:49:13 2009 From: congxin.zhao at gmail.com (congxin zhao) Date: Thu, 26 Mar 2009 09:49:13 +0800 Subject: [Freeswitch-users] help about the rtp port. In-Reply-To: <58A60265-2CBF-409A-8483-7371F82AD4B6@freeswitch.org> References: <58A60265-2CBF-409A-8483-7371F82AD4B6@freeswitch.org> Message-ID: Hi Brian, Thanks for you response. After doing research, the two endpoints(uac, uas) wants to speak with different media format, that cause the freeswitch reject the rtp flow. But could you give me an detailed explain on the meaning of nbound-bypass-media and inbound-proxy-media? Does nbound-bypass-media mean that rtp flow directly from endpoint uac to endpoin uas, and inbound-proxy-media mean that rtp flow from endpoint uacfirst arrive freeswitch, and freeswitch forward it to the endpoint uas? Thanks, -Congxin 2009/3/26 Brian West > Zero for the port means something please provide a complete sip trace. > /b > > On Mar 25, 2009, at 5:52 AM, congxin zhao wrote: > > Hi, > > I meet a issue about the rtp port. The media port value in the sdp of SIP > 200 OK which freeswitch internal sends to ua is always 0, so the media > forward is always wrong. What could cause this problem possibly? > > No matter I set > > or > > The problem always showes. > > Does the following rtp-ip and ext-rtp-ip set correct? In external.xml, I > set both rtp-ip and ext-rtp-ip as $${external_rtp_ip}, is that correct? I > am not clear about this two variables in the external.xml and internal.xml > > external.xml: > external.xml: > > internal.xml: > internal.xml: > > > Thanks, > -Congxin > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/be3c30a4/attachment.html From rjcajax at gmail.com Wed Mar 25 19:13:16 2009 From: rjcajax at gmail.com (Robert Clayton) Date: Wed, 25 Mar 2009 22:13:16 -0400 Subject: [Freeswitch-users] Lua session:setInputCallback Message-ID: All, When using Lua InputCallback while streaming audio and collecting an undetermined number of digits (finished by #) it seem that returning false, break or stop all close the InputCallback ability. I do not see the difference between the three? My question is in order to collect multiple digits the streaming must continue or be paused. Is there a way to stop the recording in such a way that digits can continue to be collected other than using pause? Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/39a2932b/attachment.html From mike at jerris.com Wed Mar 25 19:48:53 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Mar 2009 22:48:53 -0400 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <22709805.post@talk.nabble.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> <22709805.post@talk.nabble.com> Message-ID: <6A8AEA04-A2B1-46A7-9691-52D7AC791406@jerris.com> Thanks for access to your machine. The issue was that the odbc detection was trying to use odbc if either the libs or headers were found, not only if both were found. I fixed the detection to not try to use odbc if the headers were not installed. Installing unixODBC- devel package of course lets you install as normal, but we will not try to install automatically when we can't. Mike On Mar 25, 2009, at 4:01 PM, chevio wrote: > > How was this fixed ?, I am experiencing the same problem. > > Chevio > > > Shelby Ramsey-2 wrote: >> >> Thanks for the help. That did the trick. >> >> SDR >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Compile-Errors-...-tp22149989p22709805.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Wed Mar 25 21:39:31 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 26 Mar 2009 15:39:31 +1100 Subject: [Freeswitch-users] Multiple calls with PortAudio Message-ID: <20090326043931.GA6652@jdc.jasonjgw.net> This has occurred with a number of recent revisions. If anyone can reproduce it or suggest debugging steps, I'll gladly supply more information. Jira isn't convenient for me due to X issues at the moment, unless there's a way to interact with it other than via a Javascript-capable Web browser. Distribution: Debian Sid, now kernel 2.6.29 (same experience with 2.6.26). FreeSwitch revision 12701. Steps to reproduce Make a call with portAudio to a SIP endpoint (I haven't tried it with other endpoints): pa call Next, make another call: pa call Result: the party to the first call hears no music on hold, or possibly broken/intermittent music on hold, and the second call fails to go through. Where it hangs seems to vary, and sometimes it does in fact work. Often it hangs before it reaches the calling state, suggesting that there's some kind of locking issue involved. I'll perform further testing. Meanwhile, if anyone else can reproduce it, this may help. Note that the second call often doesn't return an error; it just hangs somewhere in the set-up process, e.g., init. From jason at jasonjgw.net Wed Mar 25 22:52:19 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 26 Mar 2009 16:52:19 +1100 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles In-Reply-To: <20090323235646.GA11567@jdc.jasonjgw.net> References: <20090323063100.GA5058@jdc.jasonjgw.net> <20090323235646.GA11567@jdc.jasonjgw.net> Message-ID: <20090326055219.GA7791@jdc.jasonjgw.net> Jason White wrote: > It looks like an operating system issue to me. Furthermore, the following message on linux-kernel appears relevant. http://linux.derkeiler.com/Mailing-Lists/Kernel/2005-03/3988.htm >From what I have been able to ascertain, Red Hat/Fedora kernels don't seem to give rise to this issue, whereas Debian kernels do. I haven't yet tested with a mainline kernel.org kernel. From jason at jasonjgw.net Wed Mar 25 23:25:10 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 26 Mar 2009 17:25:10 +1100 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles In-Reply-To: <20090326055219.GA7791@jdc.jasonjgw.net> References: <20090323063100.GA5058@jdc.jasonjgw.net> <20090323235646.GA11567@jdc.jasonjgw.net> <20090326055219.GA7791@jdc.jasonjgw.net> Message-ID: <20090326062510.GA8310@jdc.jasonjgw.net> I've read the ipv6(7) manual page now. Unfortunately echo 1 > /proc/sys/net/ipv6/bindv6only doesn't solve the problem as the manual page suggests it should: From moizchinoy at gmail.com Wed Mar 25 23:59:34 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 26 Mar 2009 10:59:34 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... Message-ID: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> Hi, I am having trouble compiling iksemel for google talk. There errors are in gnutls.h... I followed the instructions on: http://wiki.freeswitch.org/wiki/Dingaling http://wiki.freeswitch.org/wiki/Ixemel_MSVS_project_example GNUTLS VERSION is "2.7.3". Here are the erors: 1. Error 1 error C2061: syntax error : identifier 'gnutls_record_send' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 2. Error 2 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 3. Error 3 error C2059: syntax error : 'type' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 4. Error 4 error C2061: syntax error : identifier 'gnutls_record_recv' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 5. Error 5 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 6. Error 6 error C2059: syntax error : 'type' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 7. Error 7 error C2061: syntax error : identifier 'gnutls_record_set_max_size' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 8. Error 8 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 9. Error 9 error C2059: syntax error : 'type' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 10. Error 10 error C2143: syntax error : missing ')' before '*' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 11. Error 11 error C2143: syntax error : missing '{' before '*' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 12. Error 12 error C2059: syntax error : ')' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 13. Error 13 error C2143: syntax error : missing ')' before '*' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 14. Error 14 error C2143: syntax error : missing '{' before '*' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 15. Error 15 error C2059: syntax error : ')' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 16. Error 16 error C2146: syntax error : missing ')' before identifier 'push_func' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 17. Error 17 error C2081: 'gnutls_push_func' : name in formal parameter list illegal D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 18. Error 18 error C2061: syntax error : identifier 'push_func' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 19. Error 19 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 20. Error 20 error C2059: syntax error : ')' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 21. Error 21 error C2146: syntax error : missing ')' before identifier 'pull_func' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 22. Error 22 error C2081: 'gnutls_pull_func' : name in formal parameter list illegal D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 23. Error 23 error C2061: syntax error : identifier 'pull_func' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 24. Error 24 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 25. Error 25 error C2059: syntax error : ')' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 26. Error 26 error C2146: syntax error : missing ')' before identifier 'tls_push' d:\freeswitch-snapshot\libs\iksemel\src\stream.c 105 27. Error 27 error C2059: syntax error : ')' d:\freeswitch-snapshot\libs\iksemel\src\stream.c 105 28. Error 28 error C2146: syntax error : missing ')' before identifier 'tls_pull' d:\freeswitch-snapshot\libs\iksemel\src\stream.c 106 29. Error 29 error C2059: syntax error : ')' d:\freeswitch-snapshot\libs\iksemel\src\stream.c 106 -- Regards, Moiz Chinoy. From mike at jerris.com Thu Mar 26 01:24:18 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Mar 2009 04:24:18 -0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> Message-ID: <3BE666D9-3080-4FAD-A44B-78F230AB3EDB@jerris.com> There is no known working build on windows for the tls with freeswitch. We would be happy if someone would submit a full working build. Mike On Mar 26, 2009, at 2:59 AM, Moiz Chinoy wrote: > Hi, > > I am having trouble compiling iksemel for google talk. There errors > are in gnutls.h... > I followed the instructions on: > http://wiki.freeswitch.org/wiki/Dingaling > http://wiki.freeswitch.org/wiki/Ixemel_MSVS_project_example > > GNUTLS VERSION is "2.7.3". > > Here are the erors: > > 1. > Error 1 error C2061: syntax error : identifier > 'gnutls_record_send' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 > 2. > Error 2 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 > 3. > Error 3 error C2059: syntax error : 'type' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 > 4. > Error 4 error C2061: syntax error : identifier > 'gnutls_record_recv' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 > 5. > Error 5 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 > 6. > Error 6 error C2059: syntax error : 'type' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 > 7. > Error 7 error C2061: syntax error : identifier > 'gnutls_record_set_max_size' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 > 8. > Error 8 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 > 9. > Error 9 error C2059: syntax error : 'type' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 > 10. > Error 10 error C2143: syntax error : missing ')' before '*' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 > 11. > Error 11 error C2143: syntax error : missing '{' before '*' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 > 12. > Error 12 error C2059: syntax error : ')' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 > 13. > Error 13 error C2143: syntax error : missing ')' before '*' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 > 14. > Error 14 error C2143: syntax error : missing '{' before '*' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 > 15. > Error 15 error C2059: syntax error : ')' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 > 16. > Error 16 error C2146: syntax error : missing ')' before > identifier 'push_func' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 17. > Error 17 error C2081: 'gnutls_push_func' : name in formal > parameter list illegal > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 18. > Error 18 error C2061: syntax error : identifier 'push_func' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 19. > Error 19 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 20. > Error 20 error C2059: syntax error : ')' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 21. > Error 21 error C2146: syntax error : missing ')' before > identifier 'pull_func' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 22. > Error 22 error C2081: 'gnutls_pull_func' : name in formal > parameter list illegal > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 23. > Error 23 error C2061: syntax error : identifier 'pull_func' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 24. > Error 24 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 25. > Error 25 error C2059: syntax error : ')' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 26. > Error 26 error C2146: syntax error : missing ')' before > identifier 'tls_push' > d:\freeswitch-snapshot\libs\iksemel\src\stream.c 105 > 27. > Error 27 error C2059: syntax error : ')' > d:\freeswitch-snapshot\libs\iksemel\src\stream.c 105 > 28. > Error 28 error C2146: syntax error : missing ')' before > identifier 'tls_pull' > d:\freeswitch-snapshot\libs\iksemel\src\stream.c 106 > 29. > Error 29 error C2059: syntax error : ')' > d:\freeswitch-snapshot\libs\iksemel\src\stream.c 106 > > -- > Regards, > Moiz Chinoy. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mattdfong at gmail.com Thu Mar 26 02:40:17 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 16:40:17 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> Message-ID: <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> Hi Anthony, So it's been 2 days since my last request, so I'm due for another one ;) It would be nice if there was a way to execute a script (lua) on fifo bridge. I currently rely on the channel_bridge event, but I'm worried that as my system scales, it would be better to fire a custom event. In non-fifo mode, I can do this with bridge_pre_execute_bleg_app/data, but this doesn't work with a fifo bridge. It doesn't really matter which channel it fires on fifo out or fifo in channel, anything is better than having to listen for a specific channel_bridge on a high-use FS installation. Is there anyway to get an api/script to fire on fifo bridge currently that I'm missing? Thanks! --matt 2009/3/23 Anthony Minessale > ok, > maybe after this i can have a day off ;) > > 2 variables added to latest trunk: > > "fifo_caller_consumer_import" > "fifo_consumer_caller_import" > > both work like the regular import but one is a list of vars to copy from > caller to consumer and one is a list to copy from consumer to caller. > > > 2009/3/23 Matthew Fong > >> Thanks Anthony, for creating the transfer_after_bridge feature for me. >> Your rapid actions, are the reason I'm positive I made the right decision >> switch to to FS. >> I got another challenge to throw your way. In the current fifo >> implementation, there's no way to identify which fifo consumer, consumes a >> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >> pass data between channels when a fifo bridge is created). I want to be able >> to transfer some caller information to the consumer channel on bridge, to >> populate an agent's screen. >> >> Under normal (non-fifo) circumstances, when a call is bridged, I've used >> the 'import' channel variable, so that onBridge, the aleg automatically gets >> populated with the bleg's 'import' field. However when fifo bridges, it does >> not recognize import. In other applications, I've gotten around this by >> using bridge_pre_execute_bleg_app/data to throw a custom event but with >> fifo, bridge_pre_execute also does not work. I've been looking at the >> fifo::info event, but again, there's no fifo_action that directly links >> caller variables and consumer variables together. >> >> For now at least, I can get around this by storing uuid information in my >> separate database, and looking up the other channel's information based >> on other_leg_unique_id, but it would be nice if I could directly see it when >> the channel is bridged. Anyway, great program, and I hope you consider >> implementing these features to make FS even better. Thanks. >> >> --matt >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/f7b81407/attachment.html From asannucci at gmail.com Thu Mar 26 04:23:47 2009 From: asannucci at gmail.com (Andrea) Date: Thu, 26 Mar 2009 06:23:47 -0500 Subject: [Freeswitch-users] Error Compiling iksemel... References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> Message-ID: <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> Are you installed gnutls and gnutls-devel? Regards From moizchinoy at gmail.com Thu Mar 26 04:46:16 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 26 Mar 2009 15:46:16 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> Message-ID: <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> Only gnutls 2.7.3 Not gnutls-devel On Thu, Mar 26, 2009 at 3:23 PM, Andrea wrote: > Are you installed gnutls and gnutls-devel? > > Regards > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. From helmut.kuper at ewetel.de Thu Mar 26 07:12:13 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 26 Mar 2009 15:12:13 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions Message-ID: <49CB8D3D.7050202@ewetel.de> Hello, since a few days I observe a high CPU load of my FS server, but I have no idea what it could be. There are only a few sessions running and there is only a few log activity. 2 days ago I restarted FS, but no change. The top command shows this: top - 15:02:33 up 106 days, 30 min, 4 users, load average: 0.24, 0.35, 0.42 Tasks: 190 total, 1 running, 189 sleeping, 0 stopped, 0 zombie Cpu(s): 7.2%us, 12.2%sy, 0.0%ni, 80.0%id, 0.2%wa, 0.2%hi, 0.2%si, 0.0%st Mem: 4151776k total, 4003664k used, 148112k free, 414708k buffers Swap: 15623204k total, 88k used, 15623116k free, 2021412k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 14048 ippbx 20 0 1555m 1.2g 10m S 43 30.5 333:11.03 freeswitch 14049 ippbx 20 0 1555m 1.2g 10m S 0 30.5 5:06.88 freeswitch 14054 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:14.38 freeswitch 14055 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:58.50 freeswitch 14057 ippbx 20 0 1555m 1.2g 10m S 0 30.5 13:05.20 freeswitch 20511 ippbx 20 0 1555m 1.2g 10m S 0 30.5 0:00.14 freeswitch so only one process (PID: 14048) is causing that load. It's not the parent process (the initial FS startup process) as ps -elf shows: ippbx at ippbx-prod-node0:~/ippbx.prod$ ps -eLf | grep frees ippbx 14033 1 14033 0 28 Mar23 ? 00:00:01 bin/freeswitch -nc ippbx 14033 1 14034 0 28 Mar23 ? 00:00:08 bin/freeswitch -nc ippbx 14033 1 14035 0 28 Mar23 ? 00:03:39 bin/freeswitch -nc ippbx 14033 1 14036 0 28 Mar23 ? 00:00:07 bin/freeswitch -nc ippbx 14033 1 14037 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 14038 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 14039 0 28 Mar23 ? 00:00:02 bin/freeswitch -nc ippbx 14033 1 14042 0 28 Mar23 ? 00:03:41 bin/freeswitch -nc ippbx 14033 1 14043 0 28 Mar23 ? 00:00:03 bin/freeswitch -nc ippbx 14033 1 14044 0 28 Mar23 ? 00:00:01 bin/freeswitch -nc ippbx 14033 1 14045 0 28 Mar23 ? 00:00:25 bin/freeswitch -nc ippbx 14033 1 14046 0 28 Mar23 ? 00:01:20 bin/freeswitch -nc ippbx 14033 1 14047 0 28 Mar23 ? 00:05:32 bin/freeswitch -nc ippbx 14033 1 14048 7 28 Mar23 ? 05:33:35 bin/freeswitch -nc ippbx 14033 1 14049 0 28 Mar23 ? 00:05:07 bin/freeswitch -nc ippbx 14033 1 14050 0 28 Mar23 ? 00:01:01 bin/freeswitch -nc ippbx 14033 1 14051 0 28 Mar23 ? 00:25:43 bin/freeswitch -nc ippbx 14033 1 14052 0 28 Mar23 ? 00:00:01 bin/freeswitch -nc ippbx 14033 1 14054 0 28 Mar23 ? 00:04:14 bin/freeswitch -nc ippbx 14033 1 14055 0 28 Mar23 ? 00:04:58 bin/freeswitch -nc ippbx 14033 1 14056 0 28 Mar23 ? 00:06:23 bin/freeswitch -nc ippbx 14033 1 14057 0 28 Mar23 ? 00:13:05 bin/freeswitch -nc ippbx 14033 1 14058 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 14059 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20518 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20519 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20521 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20522 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 20526 19854 20526 0 1 15:03 pts/0 00:00:00 grep frees Doing a strace on PID 14048 prints tons of "epoll_wait(21, {}, 4, 0) = 0" lines on the screen, which eats all of my desktop pc's cpu power :/ So can a developer say what this is, or what and how should I debug to find out the cause of this? Can I shot it down via kill or "kill -9" without crashing FS totally? regards helmut From trevor at concipient.net Thu Mar 26 02:45:58 2009 From: trevor at concipient.net (Trevor Hammonds) Date: Thu, 26 Mar 2009 02:45:58 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid Message-ID: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> Has there been any progress getting FreeSWITCH to build on Ubuntu Intrepid without downgrading libtool? Thanks! Sincerely, Trevor Hammonds From Richard.Lamkin at mettoni.com Thu Mar 26 04:24:53 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 26 Mar 2009 11:24:53 -0000 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> Dear All, As a developer within a commercial organisation I would like to highlight that IRC access is blocked by my organisation. This is because it falls under the chat room category and is regarded as a security risk. Therefore is there any means of putting a digest of IRC traffic though the IRC node used for Freeswitch. I like many in the commercial world are barred by IT departmental polices from any chat room access. I feel I'm missing out on this useful stream on information. Another issue with any medium which is transitory is that I work in the UK and an therefore would not be privy to communications that occur outside my time zone. I do support efforts to put together a forum, which although less response than IRC is more permanent. I am subscribed to the User and Dev mail lists which I find are a very useful read. Regards Richard Lamkin Mettoni Group, UK ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/1bad5e3b/attachment.html From frank at impactfax.com Thu Mar 26 07:19:31 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 26 Mar 2009 10:19:31 -0400 Subject: [Freeswitch-users] compile poblem - FIXED_POINT or FLOATING_POINT Message-ID: <1de601c9ae1d$e136cf40$33014c0a@ws4> I have this version running on fedora 8 right now. Compiled fine and is in production. version FreeSWITCH Version 1.0.trunk (10960) However, I was in the src directory and ran "make current" and after starting to compile it blew up with Making all in libspeex make[4]: Entering directory `/usr/src/freeswitch/libs/speex/libspeex' /bin/sh ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c -o nb_celp.lo nb_celp.c gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c nb_celp.c -fPIC -DPIC -o nb_celp.o In file included from modes.h:41, from nb_celp.h:39, from nb_celp.c:37: arch.h:65:2: error: #error You now need to define either FIXED_POINT or FLOATING_POINT make[4]: *** [nb_celp.lo] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/libs/speex/libspeex' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/libs/speex' make[2]: *** [all] Error 2 make[2]: Leaving directory `/usr/src/freeswitch/libs/speex' make[1]: *** [libs/speex/libspeex/libspeexdsp.la] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 ----- from modules.conf- loggers/mod_console loggers/mod_logfile loggers/mod_syslog applications/mod_commands applications/mod_conference applications/mod_dptools applications/mod_enum applications/mod_fifo applications/mod_voicemail applications/mod_limit applications/mod_expr applications/mod_esf applications/mod_fsv asr_tts/mod_flite asr_tts/mod_pocketsphinx asr_tts/mod_cepstral codecs/mod_g711 codecs/mod_g723_1 codecs/mod_amr codecs/mod_g729 codecs/mod_h26x codecs/mod_voipcodecs codecs/mod_ilbc codecs/mod_speex dialplans/mod_dialplan_xml dialplans/mod_dialplan_asterisk endpoints/mod_iax endpoints/mod_sofia event_handlers/mod_event_socket event_handlers/mod_cdr_csv formats/mod_native_file formats/mod_sndfile formats/mod_local_stream formats/mod_tone_stream languages/mod_spidermonkey languages/mod_spidermonkey_teletone languages/mod_spidermonkey_core_db languages/mod_spidermonkey_socket languages/mod_lua xml_int/mod_xml_rpc xml_int/mod_xml_curl xml_int/mod_xml_cdr say/mod_say_en any insight on what might be doing on with this compile where it previously compiled fine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/f4523900/attachment-0001.html From mrene_lists at avgs.ca Thu Mar 26 07:26:10 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Mar 2009 10:26:10 -0400 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CB8D3D.7050202@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> Message-ID: <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> gcore -o fs [pid here] gdb /path/to/fs core.file thread apply all bt then look for the thread and show me the backtrace. Math On 26-Mar-09, at 10:12 AM, Helmut Kuper wrote: > Hello, > > since a few days I observe a high CPU load of my FS server, but I have > no idea what it could be. There are only a few sessions running and > there is only a few log activity. 2 days ago I restarted FS, but no > change. The top command shows this: > > top - 15:02:33 up 106 days, 30 min, 4 users, load average: 0.24, > 0.35, > 0.42 > Tasks: 190 total, 1 running, 189 sleeping, 0 stopped, 0 zombie > Cpu(s): 7.2%us, 12.2%sy, 0.0%ni, 80.0%id, 0.2%wa, 0.2%hi, 0.2%si, > 0.0%st > Mem: 4151776k total, 4003664k used, 148112k free, 414708k > buffers > Swap: 15623204k total, 88k used, 15623116k free, 2021412k > cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > 14048 ippbx 20 0 1555m 1.2g 10m S 43 30.5 333:11.03 > freeswitch > 14049 ippbx 20 0 1555m 1.2g 10m S 0 30.5 5:06.88 > freeswitch > 14054 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:14.38 > freeswitch > 14055 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:58.50 > freeswitch > 14057 ippbx 20 0 1555m 1.2g 10m S 0 30.5 13:05.20 > freeswitch > 20511 ippbx 20 0 1555m 1.2g 10m S 0 30.5 0:00.14 > freeswitch > > > so only one process (PID: 14048) is causing that load. It's not the > parent process (the initial FS startup process) as ps -elf shows: > > ippbx at ippbx-prod-node0:~/ippbx.prod$ ps -eLf | grep frees > ippbx 14033 1 14033 0 28 Mar23 ? 00:00:01 > bin/freeswitch -nc > ippbx 14033 1 14034 0 28 Mar23 ? 00:00:08 > bin/freeswitch -nc > ippbx 14033 1 14035 0 28 Mar23 ? 00:03:39 > bin/freeswitch -nc > ippbx 14033 1 14036 0 28 Mar23 ? 00:00:07 > bin/freeswitch -nc > ippbx 14033 1 14037 0 28 Mar23 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 14038 0 28 Mar23 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 14039 0 28 Mar23 ? 00:00:02 > bin/freeswitch -nc > ippbx 14033 1 14042 0 28 Mar23 ? 00:03:41 > bin/freeswitch -nc > ippbx 14033 1 14043 0 28 Mar23 ? 00:00:03 > bin/freeswitch -nc > ippbx 14033 1 14044 0 28 Mar23 ? 00:00:01 > bin/freeswitch -nc > ippbx 14033 1 14045 0 28 Mar23 ? 00:00:25 > bin/freeswitch -nc > ippbx 14033 1 14046 0 28 Mar23 ? 00:01:20 > bin/freeswitch -nc > ippbx 14033 1 14047 0 28 Mar23 ? 00:05:32 > bin/freeswitch -nc > ippbx 14033 1 14048 7 28 Mar23 ? 05:33:35 > bin/freeswitch -nc > ippbx 14033 1 14049 0 28 Mar23 ? 00:05:07 > bin/freeswitch -nc > ippbx 14033 1 14050 0 28 Mar23 ? 00:01:01 > bin/freeswitch -nc > ippbx 14033 1 14051 0 28 Mar23 ? 00:25:43 > bin/freeswitch -nc > ippbx 14033 1 14052 0 28 Mar23 ? 00:00:01 > bin/freeswitch -nc > ippbx 14033 1 14054 0 28 Mar23 ? 00:04:14 > bin/freeswitch -nc > ippbx 14033 1 14055 0 28 Mar23 ? 00:04:58 > bin/freeswitch -nc > ippbx 14033 1 14056 0 28 Mar23 ? 00:06:23 > bin/freeswitch -nc > ippbx 14033 1 14057 0 28 Mar23 ? 00:13:05 > bin/freeswitch -nc > ippbx 14033 1 14058 0 28 Mar23 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 14059 0 28 Mar23 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 20518 0 28 15:02 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 20519 0 28 15:02 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 20521 0 28 15:02 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 20522 0 28 15:02 ? 00:00:00 > bin/freeswitch -nc > ippbx 20526 19854 20526 0 1 15:03 pts/0 00:00:00 grep frees > > Doing a strace on PID 14048 prints tons of "epoll_wait(21, {}, 4, > 0) = 0" lines on the screen, which eats all of my > desktop > pc's cpu power :/ > > So can a developer say what this is, or what and how should I debug to > find out the cause of this? > Can I shot it down via kill or "kill -9" without crashing FS totally? > > regards > helmut > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mrene_lists at avgs.ca Thu Mar 26 07:26:57 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Mar 2009 10:26:57 -0400 Subject: [Freeswitch-users] compile poblem - FIXED_POINT or FLOATING_POINT In-Reply-To: <1de601c9ae1d$e136cf40$33014c0a@ws4> References: <1de601c9ae1d$e136cf40$33014c0a@ws4> Message-ID: <8E511016-5BC8-490A-9672-F1D53C580645@avgs.ca> make speex-reconf On 26-Mar-09, at 10:19 AM, Frank @ Impact wrote: > I have this version running on fedora 8 right now. Compiled fine > and is in production. > version > FreeSWITCH Version 1.0.trunk (10960) > > However, I was in the src directory and ran ?make current? and after > starting to compile it blew up with > > Making all in libspeex > make[4]: Entering directory `/usr/src/freeswitch/libs/speex/libspeex' > /bin/sh ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. > -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP > -MF .deps/nb_celp.Tpo -c -onb_celp.lo nb_celp.c > gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 - > MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c nb_celp.c -fPIC - > DPIC -o nb_celp.o > In file included from modes.h:41, > from nb_celp.h:39, > from nb_celp.c:37: > arch.h:65:2: error: #error You now need to define either FIXED_POINT > or FLOATING_POINT > make[4]: *** [nb_celp.lo] Error 1 > make[4]: Leaving directory `/usr/src/freeswitch/libs/speex/libspeex' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/src/freeswitch/libs/speex' > make[2]: *** [all] Error 2 > make[2]: Leaving directory `/usr/src/freeswitch/libs/speex' > make[1]: *** [libs/speex/libspeex/libspeexdsp.la] Error 2 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 > ----- > from modules.conf? > > loggers/mod_console > loggers/mod_logfile > loggers/mod_syslog > applications/mod_commands > applications/mod_conference > applications/mod_dptools > applications/mod_enum > applications/mod_fifo > applications/mod_voicemail > applications/mod_limit > applications/mod_expr > applications/mod_esf > applications/mod_fsv > asr_tts/mod_flite > asr_tts/mod_pocketsphinx > asr_tts/mod_cepstral > codecs/mod_g711 > codecs/mod_g723_1 > codecs/mod_amr > codecs/mod_g729 > codecs/mod_h26x > codecs/mod_voipcodecs > codecs/mod_ilbc > codecs/mod_speex > dialplans/mod_dialplan_xml > dialplans/mod_dialplan_asterisk > endpoints/mod_iax > endpoints/mod_sofia > event_handlers/mod_event_socket > event_handlers/mod_cdr_csv > formats/mod_native_file > formats/mod_sndfile > formats/mod_local_stream > formats/mod_tone_stream > languages/mod_spidermonkey > languages/mod_spidermonkey_teletone > languages/mod_spidermonkey_core_db > languages/mod_spidermonkey_socket > languages/mod_lua > xml_int/mod_xml_rpc > xml_int/mod_xml_curl > xml_int/mod_xml_cdr > say/mod_say_en > > > any insight on what might be doing on with this compile where it > previously compiled fine. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/f9980dbd/attachment-0001.html From telles-listas at devel-it.com.br Thu Mar 26 07:27:33 2009 From: telles-listas at devel-it.com.br (Rodrigo P. Telles) Date: Thu, 26 Mar 2009 11:27:33 -0300 Subject: [Freeswitch-users] Action and Anti-Action Message-ID: <49CB90D5.4090901@devel-it.com.br> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/421916b3/attachment.html From brian at freeswitch.org Thu Mar 26 07:32:32 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 09:32:32 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> Message-ID: <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> http://cgiirc.freeswitch.org/ I'm assume the web isn't blocked? /b On Mar 26, 2009, at 6:24 AM, Richard Lamkin wrote: > Dear All, > > As a developer within a commercial organisation I would like to > highlight that IRC access is blocked by my organisation. This is > because it falls under the chat room category and is regarded as a > security risk. > > Therefore is there any means of putting a digest of IRC traffic > though the IRC node used for Freeswitch. I like many in the > commercial world are barred by IT departmental polices from any chat > room access. I feel I?m missing out on this useful stream on > information. Another issue with any medium which is transitory is > that I work in the UK and an therefore would not be privy to > communications that occur outside my time zone. > > I do support efforts to put together a forum, which although less > response than IRC is more permanent. > I am subscribed to the User and Dev mail lists which I find are a > very useful read. > > Regards > > Richard Lamkin > Mettoni Group, UK > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/a3ecb8a5/attachment.html From frank at impactfax.com Thu Mar 26 07:39:46 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 26 Mar 2009 10:39:46 -0400 Subject: [Freeswitch-users] compile poblem - FIXED_POINT orFLOATING_POINT In-Reply-To: <8E511016-5BC8-490A-9672-F1D53C580645@avgs.ca> Message-ID: <1e1801c9ae20$b5b7adf0$33014c0a@ws4> Thanks. That did it. But I no longer could make mod_ilbc (and mod_flite). I had to comment it out of modules.conf because it would error with this making all mod_ilbc make[6]: *** No targets specified and no makefile found. Stop. make[5]: *** [../../../../libs/ilbc/src/libilbc.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_ilbc-all] Error 1 make[2]: *** [all-recursive] Error 1 make ilbc-reconf did not work -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Thursday, March 26, 2009 10:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] compile poblem - FIXED_POINT orFLOATING_POINT make speex-reconf On 26-Mar-09, at 10:19 AM, Frank @ Impact wrote: I have this version running on fedora 8 right now. Compiled fine and is in production. version FreeSWITCH Version 1.0.trunk (10960) However, I was in the src directory and ran "make current" and after starting to compile it blew up with Making all in libspeex make[4]: Entering directory `/usr/src/freeswitch/libs/speex/libspeex' /bin/sh ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c -onb_celp.lo nb_celp.c gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c nb_celp.c -fPIC -DPIC -o nb_celp.o In file included from modes.h:41, from nb_celp.h:39, from nb_celp.c:37: arch.h:65:2: error: #error You now need to define either FIXED_POINT or FLOATING_POINT make[4]: *** [nb_celp.lo] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/libs/speex/libspeex' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/libs/speex' make[2]: *** [all] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/073efed4/attachment-0001.html From brian at freeswitch.org Thu Mar 26 07:44:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 09:44:42 -0500 Subject: [Freeswitch-users] compile poblem - FIXED_POINT orFLOATING_POINT In-Reply-To: <1e1801c9ae20$b5b7adf0$33014c0a@ws4> References: <1e1801c9ae20$b5b7adf0$33014c0a@ws4> Message-ID: <3797D5FE-C2D8-4238-A360-C170932D1D3E@freeswitch.org> how about you just rebootstrap it all.... /b On Mar 26, 2009, at 9:39 AM, Frank @ Impact wrote: > Thanks. That did it. > But I no longer could make mod_ilbc (and mod_flite). I had to > comment it out of modules.conf because it would error with this > > making all mod_ilbc > make[6]: *** No targets specified and no makefile found. Stop. > make[5]: *** [../../../../libs/ilbc/src/libilbc.la] Error 2 > make[4]: *** [all] Error 1 > make[3]: *** [mod_ilbc-all] Error 1 > make[2]: *** [all-recursive] Error 1 > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/98ce3a3d/attachment.html From anthony.minessale at gmail.com Thu Mar 26 07:56:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 09:56:22 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> Message-ID: <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> The guy started a forum almost a month ago and as you can see nobody knows the url and it has no posts. http://freeswitch411.info/forum/ This is one of the problems I was worried about when endorsing a forum. 2009/3/26 Richard Lamkin > Dear All, > > > > As a developer within a commercial organisation I would like to highlight > that IRC access is blocked by my organisation. This is because it falls > under the chat room category and is regarded as a security risk. > > > > Therefore is there any means of putting a digest of IRC traffic though the > IRC node used for Freeswitch. I like many in the commercial world are > barred by IT departmental polices from any chat room access. I feel I?m > missing out on this useful stream on information. Another issue with any > medium which is transitory is that I work in the UK and an therefore would > not be privy to communications that occur outside my time zone. > > > > I do support efforts to put together a forum, which although less response > than IRC is more permanent. > > I am subscribed to the User and Dev mail lists which I find are a very > useful read. > > > > Regards > > > > Richard Lamkin > > Mettoni Group, UK > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/a0fb0f6c/attachment.html From anthony.minessale at gmail.com Thu Mar 26 07:59:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 09:59:05 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> Message-ID: <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> We do not support ubuntu interpid, it has at least 3 known fatal issues not experienced by all but nonetheless enough to make us unwilling to support it. It's "use at your own risk" or use the stable branch "hardy" for any support. On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds wrote: > Has there been any progress getting FreeSWITCH to build on Ubuntu > Intrepid without downgrading libtool? > > Thanks! > > Sincerely, > Trevor Hammonds > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/b4cec9c4/attachment.html From helmut.kuper at ewetel.de Thu Mar 26 08:01:53 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 26 Mar 2009 16:01:53 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> Message-ID: <49CB98E1.8080705@ewetel.de> Hi Mathieu, thx for the help :) The epoll_wait function on PID 14048 listens on fd 21 which points to "/anon_inode:[eventpoll]" On 26.03.2009 15:26, Mathieu Rene wrote: > thread apply all bt Here the output: Core was generated by `/opt/app/voip/ippbx.prod/bin/freeswitch'. [New process 14034] [New process 14035] [New process 14036] [New process 14037] [New process 14038] [New process 14039] [New process 14042] [New process 14043] [New process 14044] [New process 14045] [New process 14046] [New process 14047] [New process 14048] [New process 14049] [New process 14050] [New process 14051] [New process 14052] [New process 14054] [New process 14055] [New process 14056] [New process 14057] [New process 14058] [New process 14059] [New process 14033] #0 0xb7ee6410 in __kernel_vsyscall () (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) thread apply all bt Thread 24 (process 14033): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7d9e334 in switch_core_runtime_loop (bg=1) at src/switch_core.c:656 #5 0x0804a459 in main (argc=2, argv=0xbf9d27a4) at src/switch.c:666 Thread 23 (process 14059): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c41c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb36ef7e1 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_xml_rpc.so #3 0xb36e2612 in ChanSwitchAccept (chanSwitchP=0x81cce68, channelPP=0xab1f00e0, channelInfoPP=0xab1f00dc, errorP=0xab1f00e4) at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 #4 0xb36ee351 in ServerRun (serverP=0xb372092c) at ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 #5 0xb36df892 in mod_xml_rpc_runtime () at mod_xml_rpc.c:867 #6 0xb7da8473 in switch_loadable_module_exec (thread=0x80c38a8, obj=0x80c3698) at src/switch_loadable_module.c:94 #7 0xb7dfeeb6 in dummy_worker (opaque=0x80c38a8) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 22 (process 14058): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7d00bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7dfdbcd in apr_socket_accept (new=0xab9f134c, sock=0x82639e8, connection_context=0x66a76500) at network_io/unix/sockets.c:187 #3 0xb7d7cf4b in switch_socket_accept (new_sock=0xab9f134c, sock=0x82639e8, pool=0x66a76500) at src/switch_apr.c:664 #4 0xb328ff2c in mod_event_socket_runtime () at mod_event_socket.c:2293 #5 0xb7da8473 in switch_loadable_module_exec (thread=0x80c3640, obj=0x80c3430) at src/switch_loadable_module.c:94 #6 0xb7dfeeb6 in dummy_worker (opaque=0x80c3640) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 21 (process 14057): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=1000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7de0c25 in softtimer_runtime () at src/switch_time.c:459 #5 0xb7da8473 in switch_loadable_module_exec (thread=0x80c33d8, obj=0x80c31c8) at src/switch_loadable_module.c:94 #6 0xb7dfeeb6 in dummy_worker (opaque=0x80c33d8) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 20 (process 14056): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at locks/unix/thread_cond.c:68 #3 0xb7d7d8b4 in switch_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at src/switch_apr.c:359 #4 0xb7ddf2c8 in timer_next (timer=0xaccae0f8) at src/switch_time.c:335 #5 0xb7d8ad7c in switch_core_timer_next (timer=0x26f180a8) at src/switch_core_timer.c:76 ---Type to continue, or q to quit--- #6 0xad4e67dc in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #7 0xb7dfeeb6 in dummy_worker (opaque=0xb272b2e0) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 19 (process 14055): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at locks/unix/thread_cond.c:68 #3 0xb7d7d8b4 in switch_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at src/switch_apr.c:359 #4 0xb7ddf2c8 in timer_next (timer=0xad4af0f8) at src/switch_time.c:335 #5 0xb7d8ad7c in switch_core_timer_next (timer=0x26f180a5) at src/switch_core_timer.c:76 #6 0xad4e67dc in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #7 0xb7dfeeb6 in dummy_worker (opaque=0xb27272d0) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 18 (process 14054): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at locks/unix/thread_cond.c:68 #3 0xb7d7d8b4 in switch_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at src/switch_apr.c:359 #4 0xb7ddf2c8 in timer_next (timer=0xadf700f8) at src/switch_time.c:335 #5 0xb7d8ad7c in switch_core_timer_next (timer=0x26f180a6) at src/switch_core_timer.c:76 #6 0xad4e67dc in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #7 0xb7dfeeb6 in dummy_worker (opaque=0xb27232c0) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 17 (process 14052): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xaedcce06 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so #5 0xb7dfeeb6 in dummy_worker (opaque=0xaea4cb90) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 16 (process 14051): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb315892f in sofia_presence_event_thread_run (thread=0x80f1bb0, obj=0x0) at sofia_presence.c:664 #5 0xb7dfeeb6 in dummy_worker (opaque=0x80f1bb0) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 15 (process 14050): #0 0xb7ee6410 in __kernel_vsyscall () ---Type to continue, or q to quit--- #1 0xb7c41c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb280bf00 in wanpipe_wait (zchan=0x8146fc8, flags=0xafe0ef8c, to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 #3 0xafeecb17 in zap_channel_wait (zchan=0x64, flags=0xafe0ef8c, to=100) at src/zap_io.c:1488 #4 0xafebfb54 in zap_isdn_run (me=0x8112ab0, obj=0x811ff58) at src/ozmod/ozmod_isdn/ozmod_isdn.c:1736 #5 0xafef747a in thread_launch (args=0x8112ab0) at src/zap_threadmutex.c:74 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 14 (process 14049): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb313e306 in sofia_profile_worker_thread_run (thread=0x81004c8, obj=0x80ffb38) at sofia.c:656 #5 0xb7dfeeb6 in dummy_worker (opaque=0x81004c8) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 13 (process 14048): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c4c676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb321998d in su_epoll_port_wait_events (self=0x8107f70, tout=1000) at su_epoll_port.c:495 #3 0xb3220218 in su_base_port_run (self=0x8107f70) at su_base_port.c:349 #4 0xb321af8f in su_port_run (self=0x8107f70) at su_port.h:326 #5 0xb321af6c in su_root_run (self=0x8107770) at su_root.c:819 #6 0xb320af0a in su_pthread_port_clone_main (varg=0xb16fd098) at su_pthread_port.c:324 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 12 (process 14047): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c4c676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb321998d in su_epoll_port_wait_events (self=0x80fbaa8, tout=1000) at su_epoll_port.c:495 #3 0xb3220369 in su_base_port_step (self=0x80fbaa8, tout=1000) at su_base_port.c:467 #4 0xb321b0b5 in su_port_step (self=0x80fbaa8, tout=1000) at su_port.h:340 #5 0xb321b085 in su_root_step (self=0x80fba70, tout=1000) at su_root.c:858 #6 0xb3148442 in sofia_profile_thread_run (thread=0x81003e8, obj=0x80ffb38) at sofia.c:831 #7 0xb7dfeeb6 in dummy_worker (opaque=0x81003e8) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 11 (process 14046): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb313e306 in sofia_profile_worker_thread_run (thread=0x80f5030, obj=0x80f4280) at sofia.c:656 #5 0xb7dfeeb6 in dummy_worker (opaque=0x80f5030) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- Thread 10 (process 14045): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c4c676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb321998d in su_epoll_port_wait_events (self=0x80fd1a0, tout=1000) at su_epoll_port.c:495 #3 0xb3220218 in su_base_port_run (self=0x80fd1a0) at su_base_port.c:349 #4 0xb321af8f in su_port_run (self=0x80fd1a0) at su_port.h:326 #5 0xb321af6c in su_root_run (self=0x80fef98) at su_root.c:819 #6 0xb320af0a in su_pthread_port_clone_main (varg=0xb30ac098) at su_pthread_port.c:324 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 9 (process 14044): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c4c676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb321998d in su_epoll_port_wait_events (self=0x80e0e80, tout=1000) at su_epoll_port.c:495 #3 0xb3220369 in su_base_port_step (self=0x80e0e80, tout=1000) at su_base_port.c:467 #4 0xb321b0b5 in su_port_step (self=0x80e0e80, tout=1000) at su_port.h:340 #5 0xb321b085 in su_root_step (self=0x80f8528, tout=1000) at su_root.c:858 #6 0xb3148442 in sofia_profile_thread_run (thread=0x80f4f50, obj=0x80f4280) at sofia.c:831 #7 0xb7dfeeb6 in dummy_worker (opaque=0x80f4f50) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 8 (process 14043): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=500000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7da20a4 in switch_scheduler_task_thread (thread=0x80be590, obj=0x0) at src/switch_scheduler.c:171 #5 0xb7dfeeb6 in dummy_worker (opaque=0x80be590) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 7 (process 14042): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at locks/unix/thread_cond.c:68 #3 0xb7d7d8b4 in switch_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at src/switch_apr.c:359 #4 0xb7ddf4d6 in switch_cond_next () at src/switch_time.c:203 #5 0xb7d91b5b in switch_core_sql_thread (thread=0xb3782ae8, obj=0x0) at src/switch_core_sqldb.c:220 #6 0xb7dfeeb6 in dummy_worker (opaque=0xb3782ae8) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 6 (process 14039): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0xb70ddc60, mutex=0xb70ddc30) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0xb70ddc00, data=0xb4fe93a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0xb70ddc00, data=0xb4fe93a8) at src/switch_apr.c:879 #5 0xb7dd7bed in log_thread (t=0xb504bae0, obj=0x0) at src/switch_log.c:213 ---Type to continue, or q to quit--- #6 0xb7dfeeb6 in dummy_worker (opaque=0xb504bae0) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 5 (process 14038): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0xb7140b38, mutex=0xb7140b08) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0xb7140ad8, data=0xb58783a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0xb7140ad8, data=0xb58783a8) at src/switch_apr.c:879 #5 0xb7dae2d0 in switch_event_thread (thread=0x8068228, obj=0xb7140ad8) at src/switch_event.c:284 #6 0xb7dfeeb6 in dummy_worker (opaque=0x8068228) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 4 (process 14037): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0xb71a3b38, mutex=0xb71a3b08) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0xb71a3ad8, data=0xb60793a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0xb71a3ad8, data=0xb60793a8) at src/switch_apr.c:879 #5 0xb7dae2d0 in switch_event_thread (thread=0x8068208, obj=0xb71a3ad8) at src/switch_event.c:284 #6 0xb7dfeeb6 in dummy_worker (opaque=0x8068208) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 3 (process 14036): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x805e548, mutex=0x805e518) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0x805e4e8, data=0xb687a3a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0x805e4e8, data=0xb687a3a8) at src/switch_apr.c:879 #5 0xb7dae2d0 in switch_event_thread (thread=0x80681e8, obj=0x805e4e8) at src/switch_event.c:284 #6 0xb7dfeeb6 in dummy_worker (opaque=0x80681e8) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 2 (process 14035): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0xb70ddb38, mutex=0xb70ddb08) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0xb70ddad8, data=0xb707b3a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0xb70ddad8, data=0xb707b3a8) at src/switch_apr.c:879 #5 0xb7daf069 in switch_event_dispatch_thread (thread=0x80681c8, obj=0xb70ddad8) at src/switch_event.c:241 #6 0xb7dfeeb6 in dummy_worker (opaque=0x80681c8) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 1 (process 14034): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- #2 0xb7e00a59 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7d8d52b in pool_thread (thread=0xb79d6da8, obj=0x0) at src/switch_core_memory.c:423 #5 0xb7dfeeb6 in dummy_worker (opaque=0xb79d6da8) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 From anthony.minessale at gmail.com Thu Mar 26 08:07:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 10:07:29 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> Message-ID: <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> I'll fire 2 custom events when the call is bridged one for the consumer and one for the caller events plain custom fifo::info pull out FIFO-Name header and find the desired fifo pull out FIFO-Action header and look for bridge-consumer or bridge-caller depending on what you want to see data from. in latest trunk 2009/3/26 Matthew Fong > Hi Anthony, > So it's been 2 days since my last request, so I'm due for another one ;) > > It would be nice if there was a way to execute a script (lua) on fifo > bridge. I currently rely on the channel_bridge event, but I'm worried that > as my system scales, it would be better to fire a custom event. In non-fifo > mode, I can do this with bridge_pre_execute_bleg_app/data, but this > doesn't work with a fifo bridge. It doesn't really matter which channel it > fires on fifo out or fifo in channel, anything is better than having to > listen for a specific channel_bridge on a high-use FS installation. > > Is there anyway to get an api/script to fire on fifo bridge currently that > I'm missing? Thanks! > > --matt > > 2009/3/23 Anthony Minessale > > ok, >> maybe after this i can have a day off ;) >> >> 2 variables added to latest trunk: >> >> "fifo_caller_consumer_import" >> "fifo_consumer_caller_import" >> >> both work like the regular import but one is a list of vars to copy from >> caller to consumer and one is a list to copy from consumer to caller. >> >> >> 2009/3/23 Matthew Fong >> >>> Thanks Anthony, for creating the transfer_after_bridge feature for me. >>> Your rapid actions, are the reason I'm positive I made the right decision >>> switch to to FS. >>> I got another challenge to throw your way. In the current fifo >>> implementation, there's no way to identify which fifo consumer, consumes a >>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>> pass data between channels when a fifo bridge is created). I want to be able >>> to transfer some caller information to the consumer channel on bridge, to >>> populate an agent's screen. >>> >>> Under normal (non-fifo) circumstances, when a call is bridged, I've used >>> the 'import' channel variable, so that onBridge, the aleg automatically gets >>> populated with the bleg's 'import' field. However when fifo bridges, it does >>> not recognize import. In other applications, I've gotten around this by >>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>> fifo, bridge_pre_execute also does not work. I've been looking at the >>> fifo::info event, but again, there's no fifo_action that directly links >>> caller variables and consumer variables together. >>> >>> For now at least, I can get around this by storing uuid information in my >>> separate database, and looking up the other channel's information based >>> on other_leg_unique_id, but it would be nice if I could directly see it when >>> the channel is bridged. Anyway, great program, and I hope you consider >>> implementing these features to make FS even better. Thanks. >>> >>> --matt >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/343d3bb4/attachment.html From brian at freeswitch.org Thu Mar 26 08:07:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 10:07:56 -0500 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CB98E1.8080705@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> Message-ID: Before we go any further... what SVN rev are you on? And by heavy load what does your load average say? /b On Mar 26, 2009, at 10:01 AM, Helmut Kuper wrote: > Hi Mathieu, > > thx for the help :) > > The epoll_wait function on PID 14048 listens on fd 21 which points to > "/anon_inode:[eventpoll]" > > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/9de4eca2/attachment.html From helmut.kuper at ewetel.de Thu Mar 26 08:12:08 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 26 Mar 2009 16:12:08 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CB98E1.8080705@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> Message-ID: <49CB9B48.5030603@ewetel.de> Hi Mathieu I straced process 14047 as well and I found that it also epolls a anon_inode. But the process' epoll call lokk slightly different that 14048's: epoll_wait(13, {}, 4, 1000) = 0 Here a timeout is given. 14048 is not using a timeout (0). Maybe this helps you ... regards Helmut On 26.03.2009 16:01, Helmut Kuper wrote: > Hi Mathieu, > > thx for the help :) > > The epoll_wait function on PID 14048 listens on fd 21 which points to > "/anon_inode:[eventpoll]" From mike at jerris.com Thu Mar 26 08:16:07 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Mar 2009 11:16:07 -0400 Subject: [Freeswitch-users] Action and Anti-Action In-Reply-To: <49CB90D5.4090901@devel-it.com.br> References: <49CB90D5.4090901@devel-it.com.br> Message-ID: Actions are all run AFTER all conditions are parsed so the nated var is not set yet. you can do a single condition in this case, and set nated for use elsewhere if you need it in the actions. Mike On Mar 26, 2009, at 10:27 AM, Rodrigo P. Telles wrote: > Hi Guys, > > I'm trying to do some string matching against a created var and > looks like I am doing something wrong but I can't find whats it. > > I'm wrote an extension just for tests purposes on dialplan/ > default.xml: > > > > > > > > > > > > > > Using two SIP extensions (1000 and 1001) behind NAT and I expected > too see "Action=1" on the logs/console, but I'm seeing "Anti- > Action=1". > > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x > Execute set(nated=${cond(${network_addr} != ${sip_contact_host} ? > 1 : 0)}) > 2009-03-26 11:02:57 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1001 at x.x.x.x Expanded > String set(nated=1) > 2009-03-26 11:02:57 [DEBUG] mod_dptools.c:711 set_function() sofia/internal/1001 at x.x.x.x > SET [nated]=[1] > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x > Execute log(Anti-Action=${nated}) > 2009-03-26 11:02:57 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1001 at x.x.x.x Expanded > String log(Anti-Action=1) > 2009-03-26 11:02:57 [DEBUG] mod_dptools.c:895 log_function() Anti- > Action=1 > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x > Execute set(dialed_extension=1000) > .... > > I really appreciate any inputs. > I'm using FS 1.0.3 stable. > > regards, > Rodrigo Telles > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/7af6dc65/attachment-0001.html From mattdfong at gmail.com Thu Mar 26 08:20:09 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 22:20:09 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> Message-ID: <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> Thanks of course! But, is there any chance of firing an app? Firing an app on bridge gives the programmer more control, rather than just listening for fifo::info custom events. I find that lua running as a FS app can update my database like 10x faster than reading event_socket thru Rails/Telegraph...plus, I trust your coding much more than that of your Rail's development counterparts. :) with the custom event you are firing, you should be sure to import the variables first, then fire the event :) You rock Mr. Minessale --matt 2009/3/26 Anthony Minessale > I'll fire 2 custom events when the call is bridged one for the consumer and > one for the caller > > events plain custom fifo::info > > pull out FIFO-Name header and find the desired fifo > pull out FIFO-Action header and look for bridge-consumer or bridge-caller > depending on what you want to see data from. > > in latest trunk > > 2009/3/26 Matthew Fong > > Hi Anthony, >> So it's been 2 days since my last request, so I'm due for another one ;) >> >> It would be nice if there was a way to execute a script (lua) on fifo >> bridge. I currently rely on the channel_bridge event, but I'm worried that >> as my system scales, it would be better to fire a custom event. In non-fifo >> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >> doesn't work with a fifo bridge. It doesn't really matter which channel it >> fires on fifo out or fifo in channel, anything is better than having to >> listen for a specific channel_bridge on a high-use FS installation. >> >> Is there anyway to get an api/script to fire on fifo bridge currently that >> I'm missing? Thanks! >> >> --matt >> >> 2009/3/23 Anthony Minessale >> >> ok, >>> maybe after this i can have a day off ;) >>> >>> 2 variables added to latest trunk: >>> >>> "fifo_caller_consumer_import" >>> "fifo_consumer_caller_import" >>> >>> both work like the regular import but one is a list of vars to copy from >>> caller to consumer and one is a list to copy from consumer to caller. >>> >>> >>> 2009/3/23 Matthew Fong >>> >>>> Thanks Anthony, for creating the transfer_after_bridge feature for me. >>>> Your rapid actions, are the reason I'm positive I made the right decision >>>> switch to to FS. >>>> I got another challenge to throw your way. In the current fifo >>>> implementation, there's no way to identify which fifo consumer, consumes a >>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>> pass data between channels when a fifo bridge is created). I want to be able >>>> to transfer some caller information to the consumer channel on bridge, to >>>> populate an agent's screen. >>>> >>>> Under normal (non-fifo) circumstances, when a call is bridged, I've used >>>> the 'import' channel variable, so that onBridge, the aleg automatically gets >>>> populated with the bleg's 'import' field. However when fifo bridges, it does >>>> not recognize import. In other applications, I've gotten around this by >>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>> fifo::info event, but again, there's no fifo_action that directly links >>>> caller variables and consumer variables together. >>>> >>>> For now at least, I can get around this by storing uuid information in >>>> my separate database, and looking up the other channel's information based >>>> on other_leg_unique_id, but it would be nice if I could directly see it when >>>> the channel is bridged. Anyway, great program, and I hope you consider >>>> implementing these features to make FS even better. Thanks. >>>> >>>> --matt >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/08a73c72/attachment.html From helmut.kuper at ewetel.de Thu Mar 26 08:23:56 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 26 Mar 2009 16:23:56 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> Message-ID: <49CB9E0C.4030300@ewetel.de> Hello Brian, On 26.03.2009 16:07, Brian West wrote: > Before we go any further... what SVN rev are you on? And by heavy > load what does your load average say? I'm using "FreeSWITCH Version 1.0.trunk (12347M)" my average load is currently this: top - 16:22:00 up 106 days, 1:49, 3 users, load average: 0.64, 0.76, 0.73 FS status is: freeswitch at internal> status UP 0 years, 3 days, 0 hours, 54 minutes, 0 seconds, 407 milliseconds, 119 microseconds 2029 session(s) since startup 4 session(s) 0/30 regards helmut From mattdfong at gmail.com Thu Mar 26 08:26:38 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 22:26:38 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> Message-ID: <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> Oh, so the reason why the bridge_api_app execution is more useful, is with the custom fifo:info event, for my event_socket to read it, it has to subscribe to ALL fifo:info events, meaning I have to process fifo:info events even if they are not useful to me. With an app in lua, I can fire a custom event based on say my fifo name, this way my event_socket only has to read events for a specific fifo, rather than all fifos. it's not to make more work for u :)...although it's sort of amazing how efficient of a coder you are. --matt On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: > Thanks of course! > But, is there any chance of firing an app? Firing an app on bridge gives > the programmer more control, rather than just listening for fifo::info > custom events. I find that lua running as a FS app can update my database > like 10x faster than reading event_socket thru Rails/Telegraph...plus, I > trust your coding much more than that of your Rail's development > counterparts. :) > > with the custom event you are firing, you should be sure to import the > variables first, then fire the event :) > > You rock Mr. Minessale > > --matt > > 2009/3/26 Anthony Minessale > > I'll fire 2 custom events when the call is bridged one for the consumer and >> one for the caller >> >> events plain custom fifo::info >> >> pull out FIFO-Name header and find the desired fifo >> pull out FIFO-Action header and look for bridge-consumer or bridge-caller >> depending on what you want to see data from. >> >> in latest trunk >> >> 2009/3/26 Matthew Fong >> >> Hi Anthony, >>> So it's been 2 days since my last request, so I'm due for another one ;) >>> >>> It would be nice if there was a way to execute a script (lua) on fifo >>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>> as my system scales, it would be better to fire a custom event. In non-fifo >>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>> fires on fifo out or fifo in channel, anything is better than having to >>> listen for a specific channel_bridge on a high-use FS installation. >>> >>> Is there anyway to get an api/script to fire on fifo bridge currently >>> that I'm missing? Thanks! >>> >>> --matt >>> >>> 2009/3/23 Anthony Minessale >>> >>> ok, >>>> maybe after this i can have a day off ;) >>>> >>>> 2 variables added to latest trunk: >>>> >>>> "fifo_caller_consumer_import" >>>> "fifo_consumer_caller_import" >>>> >>>> both work like the regular import but one is a list of vars to copy from >>>> caller to consumer and one is a list to copy from consumer to caller. >>>> >>>> >>>> 2009/3/23 Matthew Fong >>>> >>>>> Thanks Anthony, for creating the transfer_after_bridge feature for >>>>> me. Your rapid actions, are the reason I'm positive I made the right >>>>> decision switch to to FS. >>>>> I got another challenge to throw your way. In the current fifo >>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>> to transfer some caller information to the consumer channel on bridge, to >>>>> populate an agent's screen. >>>>> >>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>> does not recognize import. In other applications, I've gotten around this by >>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>> caller variables and consumer variables together. >>>>> >>>>> For now at least, I can get around this by storing uuid information in >>>>> my separate database, and looking up the other channel's information based >>>>> on other_leg_unique_id, but it would be nice if I could directly see it when >>>>> the channel is bridged. Anyway, great program, and I hope you consider >>>>> implementing these features to make FS even better. Thanks. >>>>> >>>>> --matt >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/62aabebb/attachment-0001.html From mike at jerris.com Thu Mar 26 08:28:05 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Mar 2009 11:28:05 -0400 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> Message-ID: <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> You can run a lua script (at startup or manually) that creates an event consumer to do exactly what you want. Mike On Mar 26, 2009, at 11:20 AM, Matthew Fong wrote: > Thanks of course! > > But, is there any chance of firing an app? Firing an app on bridge > gives the programmer more control, rather than just listening for > fifo::info custom events. I find that lua running as a FS app can > update my database like 10x faster than reading event_socket thru > Rails/Telegraph...plus, I trust your coding much more than that of > your Rail's development counterparts. :) > > with the custom event you are firing, you should be sure to import > the variables first, then fire the event :) > > You rock Mr. Minessale > > --matt > > 2009/3/26 Anthony Minessale > I'll fire 2 custom events when the call is bridged one for the > consumer and one for the caller > > events plain custom fifo::info > > pull out FIFO-Name header and find the desired fifo > pull out FIFO-Action header and look for bridge-consumer or bridge- > caller depending on what you want to see data from. > > in latest trunk > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/280471ac/attachment.html From matt at hellohunter.com Thu Mar 26 08:30:24 2009 From: matt at hellohunter.com (Matt Hunter) Date: Thu, 26 Mar 2009 22:30:24 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> Message-ID: <4256bf830903260830y5741cad8he2374ee5e607942b@mail.gmail.com> Ahhh....can you point me to a doc or wiki, I can experiment with? --matt 2009/3/26 Michael Jerris > You can run a lua script (at startup or manually) that creates an event > consumer to do exactly what you want. > Mike > > On Mar 26, 2009, at 11:20 AM, Matthew Fong wrote: > > Thanks of course! > But, is there any chance of firing an app? Firing an app on bridge gives > the programmer more control, rather than just listening for fifo::info > custom events. I find that lua running as a FS app can update my database > like 10x faster than reading event_socket thru Rails/Telegraph...plus, I > trust your coding much more than that of your Rail's development > counterparts. :) > > with the custom event you are firing, you should be sure to import the > variables first, then fire the event :) > > You rock Mr. Minessale > > --matt > > 2009/3/26 Anthony Minessale > >> I'll fire 2 custom events when the call is bridged one for the consumer >> and one for the caller >> >> events plain custom fifo::info >> >> pull out FIFO-Name header and find the desired fifo >> pull out FIFO-Action header and look for bridge-consumer or bridge-caller >> depending on what you want to see data from. >> >> in latest trunk >> >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/5a227f84/attachment.html From anthony.minessale at gmail.com Thu Mar 26 08:38:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 10:38:54 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> Message-ID: <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> this feature is already implemented system-wide not just in fifo bridge_pre_execute_aleg_app bridge_pre_execute_aleg_data bridge_pre_execute_bleg_app bridge_pre_execute_bleg_data Set either pair of these vars (aleg is the consumer) and the application of choice would be executed right when the bridge starts. 2009/3/26 Matthew Fong > Oh, so the reason why the bridge_api_app execution is more useful, is with > the custom fifo:info event, for my event_socket to read it, it has to > subscribe to ALL fifo:info events, meaning I have to process fifo:info > events even if they are not useful to me. With an app in lua, I can fire a > custom event based on say my fifo name, this way my event_socket only has to > read events for a specific fifo, rather than all fifos. > it's not to make more work for u :)...although it's sort of amazing > how efficient of a coder you are. > > --matt > > > On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: > >> Thanks of course! >> But, is there any chance of firing an app? Firing an app on bridge gives >> the programmer more control, rather than just listening for fifo::info >> custom events. I find that lua running as a FS app can update my database >> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >> trust your coding much more than that of your Rail's development >> counterparts. :) >> >> with the custom event you are firing, you should be sure to import the >> variables first, then fire the event :) >> >> You rock Mr. Minessale >> >> --matt >> >> 2009/3/26 Anthony Minessale >> >> I'll fire 2 custom events when the call is bridged one for the consumer >>> and one for the caller >>> >>> events plain custom fifo::info >>> >>> pull out FIFO-Name header and find the desired fifo >>> pull out FIFO-Action header and look for bridge-consumer or bridge-caller >>> depending on what you want to see data from. >>> >>> in latest trunk >>> >>> 2009/3/26 Matthew Fong >>> >>> Hi Anthony, >>>> So it's been 2 days since my last request, so I'm due for another one ;) >>>> >>>> It would be nice if there was a way to execute a script (lua) on fifo >>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>> fires on fifo out or fifo in channel, anything is better than having to >>>> listen for a specific channel_bridge on a high-use FS installation. >>>> >>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>> that I'm missing? Thanks! >>>> >>>> --matt >>>> >>>> 2009/3/23 Anthony Minessale >>>> >>>> ok, >>>>> maybe after this i can have a day off ;) >>>>> >>>>> 2 variables added to latest trunk: >>>>> >>>>> "fifo_caller_consumer_import" >>>>> "fifo_consumer_caller_import" >>>>> >>>>> both work like the regular import but one is a list of vars to copy >>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>> >>>>> >>>>> 2009/3/23 Matthew Fong >>>>> >>>>>> Thanks Anthony, for creating the transfer_after_bridge feature for >>>>>> me. Your rapid actions, are the reason I'm positive I made the right >>>>>> decision switch to to FS. >>>>>> I got another challenge to throw your way. In the current fifo >>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>> populate an agent's screen. >>>>>> >>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>> caller variables and consumer variables together. >>>>>> >>>>>> For now at least, I can get around this by storing uuid information in >>>>>> my separate database, and looking up the other channel's information based >>>>>> on other_leg_unique_id, but it would be nice if I could directly see it when >>>>>> the channel is bridged. Anyway, great program, and I hope you consider >>>>>> implementing these features to make FS even better. Thanks. >>>>>> >>>>>> --matt >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/22b574ae/attachment-0001.html From diego.viola at gmail.com Thu Mar 26 08:46:39 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 26 Mar 2009 11:46:39 -0400 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> Message-ID: <86a32abc0903260846t3f2390f0sa7dcb8c0230aa4df@mail.gmail.com> Use what Brian said or ssh into some Linux box and use irssi. Diego 2009/3/26 Brian West > http://cgiirc.freeswitch.org/ > I'm assume the web isn't blocked? > > /b > > On Mar 26, 2009, at 6:24 AM, Richard Lamkin wrote: > > Dear All, > > As a developer within a commercial organisation I would like to highlight > that IRC access is blocked by my organisation. This is because it falls > under the chat room category and is regarded as a security risk. > > Therefore is there any means of putting a digest of IRC traffic though the > IRC node used for Freeswitch. I like many in the commercial world are > barred by IT departmental polices from any chat room access. I feel I?m > missing out on this useful stream on information. Another issue with any > medium which is transitory is that I work in the UK and an therefore would > not be privy to communications that occur outside my time zone. > > I do support efforts to put together a forum, which although less response > than IRC is more permanent. > I am subscribed to the User and Dev mail lists which I find are a very > useful read. > > Regards > > Richard Lamkin > Mettoni Group, UK > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/eda63ebf/attachment.html From matt at hellohunter.com Thu Mar 26 08:52:50 2009 From: matt at hellohunter.com (Matt Hunter) Date: Thu, 26 Mar 2009 22:52:50 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> Message-ID: <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> Ooooo, then this is an error. I'm using FreeSWITCH Version 1.0.trunk (12701M) and setting on both the fifo in and fifo out channels, but aleg.lua/bleg.lua never get executed on fifo bridge. Do you need a trace or anything? --matt 2009/3/26 Anthony Minessale > this feature is already implemented system-wide not just in fifo > > bridge_pre_execute_aleg_app > bridge_pre_execute_aleg_data > > bridge_pre_execute_bleg_app > bridge_pre_execute_bleg_data > > Set either pair of these vars (aleg is the consumer) > > and the application of choice would be executed right when the bridge > starts. > > > > 2009/3/26 Matthew Fong > >> Oh, so the reason why the bridge_api_app execution is more useful, is with >> the custom fifo:info event, for my event_socket to read it, it has to >> subscribe to ALL fifo:info events, meaning I have to process fifo:info >> events even if they are not useful to me. With an app in lua, I can fire a >> custom event based on say my fifo name, this way my event_socket only has to >> read events for a specific fifo, rather than all fifos. >> it's not to make more work for u :)...although it's sort of amazing >> how efficient of a coder you are. >> >> --matt >> >> >> On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: >> >>> Thanks of course! >>> But, is there any chance of firing an app? Firing an app on bridge gives >>> the programmer more control, rather than just listening for fifo::info >>> custom events. I find that lua running as a FS app can update my database >>> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >>> trust your coding much more than that of your Rail's development >>> counterparts. :) >>> >>> with the custom event you are firing, you should be sure to import the >>> variables first, then fire the event :) >>> >>> You rock Mr. Minessale >>> >>> --matt >>> >>> 2009/3/26 Anthony Minessale >>> >>> I'll fire 2 custom events when the call is bridged one for the consumer >>>> and one for the caller >>>> >>>> events plain custom fifo::info >>>> >>>> pull out FIFO-Name header and find the desired fifo >>>> pull out FIFO-Action header and look for bridge-consumer or >>>> bridge-caller depending on what you want to see data from. >>>> >>>> in latest trunk >>>> >>>> 2009/3/26 Matthew Fong >>>> >>>> Hi Anthony, >>>>> So it's been 2 days since my last request, so I'm due for another one >>>>> ;) >>>>> >>>>> It would be nice if there was a way to execute a script (lua) on fifo >>>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>>> fires on fifo out or fifo in channel, anything is better than having to >>>>> listen for a specific channel_bridge on a high-use FS installation. >>>>> >>>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>>> that I'm missing? Thanks! >>>>> >>>>> --matt >>>>> >>>>> 2009/3/23 Anthony Minessale >>>>> >>>>> ok, >>>>>> maybe after this i can have a day off ;) >>>>>> >>>>>> 2 variables added to latest trunk: >>>>>> >>>>>> "fifo_caller_consumer_import" >>>>>> "fifo_consumer_caller_import" >>>>>> >>>>>> both work like the regular import but one is a list of vars to copy >>>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>>> >>>>>> >>>>>> 2009/3/23 Matthew Fong >>>>>> >>>>>>> Thanks Anthony, for creating the transfer_after_bridge feature for >>>>>>> me. Your rapid actions, are the reason I'm positive I made the right >>>>>>> decision switch to to FS. >>>>>>> I got another challenge to throw your way. In the current fifo >>>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>>> populate an agent's screen. >>>>>>> >>>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>>> caller variables and consumer variables together. >>>>>>> >>>>>>> For now at least, I can get around this by storing uuid information >>>>>>> in my separate database, and looking up the other channel's information >>>>>>> based on other_leg_unique_id, but it would be nice if I could directly see >>>>>>> it when the channel is bridged. Anyway, great program, and I hope you >>>>>>> consider implementing these features to make FS even better. Thanks. >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/5d529dd7/attachment-0001.html From mattdfong at gmail.com Thu Mar 26 08:53:49 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 22:53:49 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> Message-ID: <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> Woops, my double identity of my marketing alias isn't subscribed correctly...------------- Ooooo, then this is an error because bridge_pre_execute_aleg is not firing on fifo bridge. I'm using FreeSWITCH Version 1.0.trunk (12701M) and setting on both the fifo in and fifo out channels, but aleg.lua/bleg.lua never get executed on fifo bridge. Do you need a trace or anything? --matt On Thu, Mar 26, 2009 at 10:52 PM, Matt Hunter wrote: > > > 2009/3/26 Anthony Minessale > >> this feature is already implemented system-wide not just in fifo >> >> bridge_pre_execute_aleg_app >> bridge_pre_execute_aleg_data >> >> bridge_pre_execute_bleg_app >> bridge_pre_execute_bleg_data >> >> Set either pair of these vars (aleg is the consumer) >> >> and the application of choice would be executed right when the bridge >> starts. >> >> >> >> 2009/3/26 Matthew Fong >> >>> Oh, so the reason why the bridge_api_app execution is more useful, is >>> with the custom fifo:info event, for my event_socket to read it, it has to >>> subscribe to ALL fifo:info events, meaning I have to process fifo:info >>> events even if they are not useful to me. With an app in lua, I can fire a >>> custom event based on say my fifo name, this way my event_socket only has to >>> read events for a specific fifo, rather than all fifos. >>> it's not to make more work for u :)...although it's sort of amazing >>> how efficient of a coder you are. >>> >>> --matt >>> >>> >>> On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: >>> >>>> Thanks of course! >>>> But, is there any chance of firing an app? Firing an app on bridge gives >>>> the programmer more control, rather than just listening for fifo::info >>>> custom events. I find that lua running as a FS app can update my database >>>> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >>>> trust your coding much more than that of your Rail's development >>>> counterparts. :) >>>> >>>> with the custom event you are firing, you should be sure to import the >>>> variables first, then fire the event :) >>>> >>>> You rock Mr. Minessale >>>> >>>> --matt >>>> >>>> 2009/3/26 Anthony Minessale >>>> >>>> I'll fire 2 custom events when the call is bridged one for the consumer >>>>> and one for the caller >>>>> >>>>> events plain custom fifo::info >>>>> >>>>> pull out FIFO-Name header and find the desired fifo >>>>> pull out FIFO-Action header and look for bridge-consumer or >>>>> bridge-caller depending on what you want to see data from. >>>>> >>>>> in latest trunk >>>>> >>>>> 2009/3/26 Matthew Fong >>>>> >>>>> Hi Anthony, >>>>>> So it's been 2 days since my last request, so I'm due for another one >>>>>> ;) >>>>>> >>>>>> It would be nice if there was a way to execute a script (lua) on fifo >>>>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>>>> fires on fifo out or fifo in channel, anything is better than having to >>>>>> listen for a specific channel_bridge on a high-use FS installation. >>>>>> >>>>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>>>> that I'm missing? Thanks! >>>>>> >>>>>> --matt >>>>>> >>>>>> 2009/3/23 Anthony Minessale >>>>>> >>>>>> ok, >>>>>>> maybe after this i can have a day off ;) >>>>>>> >>>>>>> 2 variables added to latest trunk: >>>>>>> >>>>>>> "fifo_caller_consumer_import" >>>>>>> "fifo_consumer_caller_import" >>>>>>> >>>>>>> both work like the regular import but one is a list of vars to copy >>>>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>>>> >>>>>>> >>>>>>> 2009/3/23 Matthew Fong >>>>>>> >>>>>>>> Thanks Anthony, for creating the transfer_after_bridge feature for >>>>>>>> me. Your rapid actions, are the reason I'm positive I made the right >>>>>>>> decision switch to to FS. >>>>>>>> I got another challenge to throw your way. In the current fifo >>>>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>>>> populate an agent's screen. >>>>>>>> >>>>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>>>> caller variables and consumer variables together. >>>>>>>> >>>>>>>> For now at least, I can get around this by storing uuid information >>>>>>>> in my separate database, and looking up the other channel's information >>>>>>>> based on other_leg_unique_id, but it would be nice if I could directly see >>>>>>>> it when the channel is bridged. Anyway, great program, and I hope you >>>>>>>> consider implementing these features to make FS even better. Thanks. >>>>>>>> >>>>>>>> --matt >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> >>>>>>> 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:213-799-1400 >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/38f794a2/attachment.html From brian at freeswitch.org Thu Mar 26 08:33:43 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 10:33:43 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260830y5741cad8he2374ee5e607942b@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> <4256bf830903260830y5741cad8he2374ee5e607942b@mail.gmail.com> Message-ID: <9A65047C-F6E8-4298-A4B9-66193159C619@freeswitch.org> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.EventConsumer On Mar 26, 2009, at 10:30 AM, Matt Hunter wrote: > Ahhh....can you point me to a doc or wiki, I can experiment with? > From mattdfong at gmail.com Thu Mar 26 09:00:23 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 23:00:23 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <9A65047C-F6E8-4298-A4B9-66193159C619@freeswitch.org> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> <4256bf830903260830y5741cad8he2374ee5e607942b@mail.gmail.com> <9A65047C-F6E8-4298-A4B9-66193159C619@freeswitch.org> Message-ID: <4256bf830903260900l7d152800p9678663b4abe28ed@mail.gmail.com> Hi Brian, Thanks for the link...I saw that, but i'm a newbie to lua (only use it cause of FS), and I'm a little confused how the example works. It consumes all events? Then subscribes to a session? and then, every second checks to see if an event has been fired for that session? Would it be possible to get an idea of how to subscribe to all events, and have a function execute for each time an event is fired? Can lua "wait" until an event is fired, or must it loop and sleep every second? Thanks for the help. --matt --- the example... con = freeswitch.EventConsumer("all"); session = freeswitch.Session("sofia/default/dest at host.com"); while session:ready() do session:execute("sleep", "1000"); for e in (function() return con:pop() end) do print("event\n" .. e:serialize("xml")); end end 2009/3/26 Brian West > http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.EventConsumer > > On Mar 26, 2009, at 10:30 AM, Matt Hunter wrote: > > > Ahhh....can you point me to a doc or wiki, I can experiment with? > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/b96eb8c5/attachment.html From timr at asteriasgi.com Thu Mar 26 09:01:15 2009 From: timr at asteriasgi.com (Tim Ringenbach) Date: Thu, 26 Mar 2009 11:01:15 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> Message-ID: <49CBA6CB.4030005@asteriasgi.com> Is there nothing out there that integrates a forum with a mailing list? It seems like one could display the mailing list archives exactly like a forum, and allow users to register to the forum and post (appearing to the mailing list as username at forumurl.org) in such a way that they don't have to realize it's a mailing list. Anthony Minessale wrote: > The guy started a forum almost a month ago and as you can see nobody > knows the url and it has no posts. > > http://freeswitch411.info/forum/ > > This is one of the problems I was worried about when endorsing a forum. > From anthony.minessale at gmail.com Thu Mar 26 09:04:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 11:04:11 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> Message-ID: <191c3a030903260904m466e10c0r3df8c38ee690b4ad@mail.gmail.com> see below 2009/3/26 Matthew Fong > Woops, my double identity of my marketing alias isn't subscribed > correctly...------------- > > Ooooo, then this is an error because bridge_pre_execute_aleg is not firing > on fifo bridge. I'm using > FreeSWITCH Version 1.0.trunk (12701M) > > and setting > > > data="bridge_pre_execute_aleg_app=aleg.lua"/> <-- aleg_data not aleg_app > again...... > > data="bridge_pre_execute_bleg_app=bleg.lua"/> <-- bleg_data not bleg_app > again > > on both the fifo in and fifo out channels, but aleg.lua/bleg.lua never get > executed on fifo bridge. Do you need a trace or anything? > > --matt > On Thu, Mar 26, 2009 at 10:52 PM, Matt Hunter wrote: > >> >> >> 2009/3/26 Anthony Minessale >> >>> this feature is already implemented system-wide not just in fifo >>> >>> bridge_pre_execute_aleg_app >>> bridge_pre_execute_aleg_data >>> >>> bridge_pre_execute_bleg_app >>> bridge_pre_execute_bleg_data >>> >>> Set either pair of these vars (aleg is the consumer) >>> >>> and the application of choice would be executed right when the bridge >>> starts. >>> >>> >>> >>> 2009/3/26 Matthew Fong >>> >>>> Oh, so the reason why the bridge_api_app execution is more useful, is >>>> with the custom fifo:info event, for my event_socket to read it, it has to >>>> subscribe to ALL fifo:info events, meaning I have to process fifo:info >>>> events even if they are not useful to me. With an app in lua, I can fire a >>>> custom event based on say my fifo name, this way my event_socket only has to >>>> read events for a specific fifo, rather than all fifos. >>>> it's not to make more work for u :)...although it's sort of amazing >>>> how efficient of a coder you are. >>>> >>>> --matt >>>> >>>> >>>> On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: >>>> >>>>> Thanks of course! >>>>> But, is there any chance of firing an app? Firing an app on bridge >>>>> gives the programmer more control, rather than just listening for fifo::info >>>>> custom events. I find that lua running as a FS app can update my database >>>>> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >>>>> trust your coding much more than that of your Rail's development >>>>> counterparts. :) >>>>> >>>>> with the custom event you are firing, you should be sure to import the >>>>> variables first, then fire the event :) >>>>> >>>>> You rock Mr. Minessale >>>>> >>>>> --matt >>>>> >>>>> 2009/3/26 Anthony Minessale >>>>> >>>>> I'll fire 2 custom events when the call is bridged one for the consumer >>>>>> and one for the caller >>>>>> >>>>>> events plain custom fifo::info >>>>>> >>>>>> pull out FIFO-Name header and find the desired fifo >>>>>> pull out FIFO-Action header and look for bridge-consumer or >>>>>> bridge-caller depending on what you want to see data from. >>>>>> >>>>>> in latest trunk >>>>>> >>>>>> 2009/3/26 Matthew Fong >>>>>> >>>>>> Hi Anthony, >>>>>>> So it's been 2 days since my last request, so I'm due for another one >>>>>>> ;) >>>>>>> >>>>>>> It would be nice if there was a way to execute a script (lua) on fifo >>>>>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>>>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>>>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>>>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>>>>> fires on fifo out or fifo in channel, anything is better than having to >>>>>>> listen for a specific channel_bridge on a high-use FS installation. >>>>>>> >>>>>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>>>>> that I'm missing? Thanks! >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> 2009/3/23 Anthony Minessale >>>>>>> >>>>>>> ok, >>>>>>>> maybe after this i can have a day off ;) >>>>>>>> >>>>>>>> 2 variables added to latest trunk: >>>>>>>> >>>>>>>> "fifo_caller_consumer_import" >>>>>>>> "fifo_consumer_caller_import" >>>>>>>> >>>>>>>> both work like the regular import but one is a list of vars to copy >>>>>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>>>>> >>>>>>>> >>>>>>>> 2009/3/23 Matthew Fong >>>>>>>> >>>>>>>>> Thanks Anthony, for creating the transfer_after_bridge feature >>>>>>>>> for me. Your rapid actions, are the reason I'm positive I made the right >>>>>>>>> decision switch to to FS. >>>>>>>>> I got another challenge to throw your way. In the current fifo >>>>>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>>>>> populate an agent's screen. >>>>>>>>> >>>>>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>>>>> caller variables and consumer variables together. >>>>>>>>> >>>>>>>>> For now at least, I can get around this by storing uuid information >>>>>>>>> in my separate database, and looking up the other channel's information >>>>>>>>> based on other_leg_unique_id, but it would be nice if I could directly see >>>>>>>>> it when the channel is bridged. Anyway, great program, and I hope you >>>>>>>>> consider implementing these features to make FS even better. Thanks. >>>>>>>>> >>>>>>>>> --matt >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II >>>>>>>> >>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>> >>>>>>>> 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:213-799-1400 >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/cd092047/attachment-0001.html From Richard.Lamkin at mettoni.com Thu Mar 26 09:05:16 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 26 Mar 2009 16:05:16 -0000 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com><3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804B2AF06@nickel.mettonigroup.com> Brian, Thank you this worked for me; I have put these details in the Freeswitch wiki IRC page Regards Richard Lamkin From: Brian West [mailto:brian at freeswitch.org] Sent: 26 March 2009 14:33 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] IRC is not for all http://cgiirc.freeswitch.org/ I'm assume the web isn't blocked? /b On Mar 26, 2009, at 6:24 AM, Richard Lamkin wrote: Dear All, As a developer within a commercial organisation I would like to highlight that IRC access is blocked by my organisation. This is because it falls under the chat room category and is regarded as a security risk. Therefore is there any means of putting a digest of IRC traffic though the IRC node used for Freeswitch. I like many in the commercial world are barred by IT departmental polices from any chat room access. I feel I'm missing out on this useful stream on information. Another issue with any medium which is transitory is that I work in the UK and an therefore would not be privy to communications that occur outside my time zone. I do support efforts to put together a forum, which although less response than IRC is more permanent. I am subscribed to the User and Dev mail lists which I find are a very useful read. Regards Richard Lamkin Mettoni Group, UK ************************************************************************ * 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************ * _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/2fe3d4a4/attachment.html From anthony.minessale at gmail.com Thu Mar 26 09:05:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 11:05:19 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <49CBA6CB.4030005@asteriasgi.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> Message-ID: <191c3a030903260905y389dadc0mb1c55636ce87dbf9@mail.gmail.com> That is our plan (nabble does this) On Thu, Mar 26, 2009 at 11:01 AM, Tim Ringenbach wrote: > Is there nothing out there that integrates a forum with a mailing list? > It seems like one could display the mailing list archives exactly like a > forum, and allow users to register to the forum and post (appearing to > the mailing list as username at forumurl.org) in such a way that they don't > have to realize it's a mailing list. > > Anthony Minessale wrote: > > The guy started a forum almost a month ago and as you can see nobody > > knows the url and it has no posts. > > > > http://freeswitch411.info/forum/ > > > > This is one of the problems I was worried about when endorsing a forum. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/9acb255e/attachment.html From mike at jerris.com Thu Mar 26 09:06:54 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Mar 2009 12:06:54 -0400 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <49CBA6CB.4030005@asteriasgi.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> Message-ID: http://n2.nabble.com/freeswitch-users-f2379917.html Mike On Mar 26, 2009, at 12:01 PM, Tim Ringenbach wrote: > Is there nothing out there that integrates a forum with a mailing > list? > It seems like one could display the mailing list archives exactly > like a > forum, and allow users to register to the forum and post (appearing to > the mailing list as username at forumurl.org) in such a way that they > don't > have to realize it's a mailing list. > > Anthony Minessale wrote: >> The guy started a forum almost a month ago and as you can see nobody >> knows the url and it has no posts. >> >> http://freeswitch411.info/forum/ >> >> This is one of the problems I was worried about when endorsing a >> forum. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Thu Mar 26 09:07:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Mar 2009 09:07:59 -0700 Subject: [Freeswitch-users] Action and Anti-Action In-Reply-To: <49CB90D5.4090901@devel-it.com.br> References: <49CB90D5.4090901@devel-it.com.br> Message-ID: <87f2f3b90903260907i3568f8edt9794ad852c514656@mail.gmail.com> Look in the default.xml dialplan file for the "tod_example" extension. (It's near the top of the file.) It has an example of how to create an extension that simply sets a variable that can be used in other extensions in the dialplan. -MC 2009/3/26 Rodrigo P. Telles : > Hi Guys, > > I'm trying to do some string matching against a created var and looks like I > am doing something wrong but I can't find whats it. > > I'm wrote an extension just for tests purposes on dialplan/default.xml: > > > ???? > ??????? > ???? > ???? > ?????? > ?????? > ?????? > ?????? > ???? > > > Using two SIP extensions (1000 and 1001) behind NAT and I expected too see > "Action=1" on the logs/console, but I'm seeing "Anti-Action=1". > > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x Execute > set(nated=${cond(${network_addr} != ${sip_contact_host} ? 1 : 0)}) > 2009-03-26 11:02:57 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1001 at x.x.x.x Expanded String > set(nated=1) > 2009-03-26 11:02:57 [DEBUG] mod_dptools.c:711 set_function() > sofia/internal/1001 at x.x.x.x SET [nated]=[1] > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x Execute > log(Anti-Action=${nated}) > 2009-03-26 11:02:57 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1001 at x.x.x.x Expanded String > log(Anti-Action=1) > 2009-03-26 11:02:57 [DEBUG] mod_dptools.c:895 log_function() Anti-Action=1 > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x Execute > set(dialed_extension=1000) > .... > > I really appreciate any inputs. > I'm using FS 1.0.3 stable. > > regards, > Rodrigo Telles > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Mar 26 09:13:54 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 11:13:54 -0500 Subject: [Freeswitch-users] Action and Anti-Action In-Reply-To: <87f2f3b90903260907i3568f8edt9794ad852c514656@mail.gmail.com> References: <49CB90D5.4090901@devel-it.com.br> <87f2f3b90903260907i3568f8edt9794ad852c514656@mail.gmail.com> Message-ID: Remember the dialplan is parsed, installed into the session then sent into the execute state. The dialplan is NOT execute line by line as it goes thru the dialplan. Which means you can not set a var on one line then condition on that var on the next line because the var isn't set yet... Remember its like a todo list one by one its compiled but not executed yet. This is usually an alien concept when you're migrating from various other telephony software. /b On Mar 26, 2009, at 11:07 AM, Michael Collins wrote: > Look in the default.xml dialplan file for the "tod_example" extension. > (It's near the top of the file.) It has an example of how to create an > extension that simply sets a variable that can be used in other > extensions in the dialplan. > > -MC Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/f4c195f1/attachment.html From mattdfong at gmail.com Thu Mar 26 09:15:59 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 23:15:59 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> Message-ID: <4256bf830903260915g6c47f256k55aab73ddc8a9f25@mail.gmail.com> Yah, your right. it works...but it must be set on the fifo out channel (consumer's channel), it will not execute if it's set on the fifo in channel (caller's channel). Also api_after_bridge does not execute...but as long as bridge_pre_execute_a/bleg works, I'm super happy. Thanks. --matt On Thu, Mar 26, 2009 at 10:53 PM, Matthew Fong wrote: > Woops, my double identity of my marketing alias isn't subscribed > correctly...------------- > > Ooooo, then this is an error because bridge_pre_execute_aleg is not firing > on fifo bridge. I'm using > FreeSWITCH Version 1.0.trunk (12701M) > > and setting > > > data="bridge_pre_execute_aleg_app=aleg.lua"/> > > data="bridge_pre_execute_bleg_app=bleg.lua"/> > > on both the fifo in and fifo out channels, but aleg.lua/bleg.lua never get > executed on fifo bridge. Do you need a trace or anything? > > --matt > On Thu, Mar 26, 2009 at 10:52 PM, Matt Hunter wrote: > >> >> >> 2009/3/26 Anthony Minessale >> >>> this feature is already implemented system-wide not just in fifo >>> >>> bridge_pre_execute_aleg_app >>> bridge_pre_execute_aleg_data >>> >>> bridge_pre_execute_bleg_app >>> bridge_pre_execute_bleg_data >>> >>> Set either pair of these vars (aleg is the consumer) >>> >>> and the application of choice would be executed right when the bridge >>> starts. >>> >>> >>> >>> 2009/3/26 Matthew Fong >>> >>>> Oh, so the reason why the bridge_api_app execution is more useful, is >>>> with the custom fifo:info event, for my event_socket to read it, it has to >>>> subscribe to ALL fifo:info events, meaning I have to process fifo:info >>>> events even if they are not useful to me. With an app in lua, I can fire a >>>> custom event based on say my fifo name, this way my event_socket only has to >>>> read events for a specific fifo, rather than all fifos. >>>> it's not to make more work for u :)...although it's sort of amazing >>>> how efficient of a coder you are. >>>> >>>> --matt >>>> >>>> >>>> On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: >>>> >>>>> Thanks of course! >>>>> But, is there any chance of firing an app? Firing an app on bridge >>>>> gives the programmer more control, rather than just listening for fifo::info >>>>> custom events. I find that lua running as a FS app can update my database >>>>> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >>>>> trust your coding much more than that of your Rail's development >>>>> counterparts. :) >>>>> >>>>> with the custom event you are firing, you should be sure to import the >>>>> variables first, then fire the event :) >>>>> >>>>> You rock Mr. Minessale >>>>> >>>>> --matt >>>>> >>>>> 2009/3/26 Anthony Minessale >>>>> >>>>> I'll fire 2 custom events when the call is bridged one for the consumer >>>>>> and one for the caller >>>>>> >>>>>> events plain custom fifo::info >>>>>> >>>>>> pull out FIFO-Name header and find the desired fifo >>>>>> pull out FIFO-Action header and look for bridge-consumer or >>>>>> bridge-caller depending on what you want to see data from. >>>>>> >>>>>> in latest trunk >>>>>> >>>>>> 2009/3/26 Matthew Fong >>>>>> >>>>>> Hi Anthony, >>>>>>> So it's been 2 days since my last request, so I'm due for another one >>>>>>> ;) >>>>>>> >>>>>>> It would be nice if there was a way to execute a script (lua) on fifo >>>>>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>>>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>>>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>>>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>>>>> fires on fifo out or fifo in channel, anything is better than having to >>>>>>> listen for a specific channel_bridge on a high-use FS installation. >>>>>>> >>>>>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>>>>> that I'm missing? Thanks! >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> 2009/3/23 Anthony Minessale >>>>>>> >>>>>>> ok, >>>>>>>> maybe after this i can have a day off ;) >>>>>>>> >>>>>>>> 2 variables added to latest trunk: >>>>>>>> >>>>>>>> "fifo_caller_consumer_import" >>>>>>>> "fifo_consumer_caller_import" >>>>>>>> >>>>>>>> both work like the regular import but one is a list of vars to copy >>>>>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>>>>> >>>>>>>> >>>>>>>> 2009/3/23 Matthew Fong >>>>>>>> >>>>>>>>> Thanks Anthony, for creating the transfer_after_bridge feature >>>>>>>>> for me. Your rapid actions, are the reason I'm positive I made the right >>>>>>>>> decision switch to to FS. >>>>>>>>> I got another challenge to throw your way. In the current fifo >>>>>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>>>>> populate an agent's screen. >>>>>>>>> >>>>>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>>>>> caller variables and consumer variables together. >>>>>>>>> >>>>>>>>> For now at least, I can get around this by storing uuid information >>>>>>>>> in my separate database, and looking up the other channel's information >>>>>>>>> based on other_leg_unique_id, but it would be nice if I could directly see >>>>>>>>> it when the channel is bridged. Anyway, great program, and I hope you >>>>>>>>> consider implementing these features to make FS even better. Thanks. >>>>>>>>> >>>>>>>>> --matt >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II >>>>>>>> >>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>> >>>>>>>> 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:213-799-1400 >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/006e694a/attachment-0001.html From jonas.gauffin at gmail.com Thu Mar 26 11:09:55 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Thu, 26 Mar 2009 19:09:55 +0100 Subject: [Freeswitch-users] sipp emulating a registered end point Message-ID: Hello I want to achive this: Sipp1 -> FS -> Sipp2 Sipp1 emulates a inbound calls (easy to achive) Sipp2 should emulate a registered user (i.e. register with FS and then just wait for calls and hangup when sipp1 hangsup) How do I configure sipp as "Sipp2"? Thanks, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/12d72e84/attachment.html From Richard.Lamkin at mettoni.com Thu Mar 26 11:26:42 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 26 Mar 2009 18:26:42 -0000 Subject: [Freeswitch-users] Load testing and thread use Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> Dear All, I am testing FS as a call queuing server. My test set up is as follows; Two Windows XP (SP3) Pc's with 1.0.3 installed out of the box. [ Before anyone says use Linux I need to use windows for a specific reason] I have a single gateway [fsb1500] on FS-1 configured to register with FS-2 [extn 1500]. I then use the following CLI command to create a call from FS1[5900] to FS2[5900] "bgapi originate sofia/gateway/fsb1500/5900 at richardl-5013-2.mettonigroup.com 5900" This happily connects FS1[5900] to FS2[5900] . Codec set to G711a. I have one hardphone on FS2 with a call connected to 5900 to allow me to listen to MOH. I have been monitoring the system resources used by FS and have observed that for each new call a new thread is created. I have pushed well over 200 calls from FS-1 into FS2. All goes well with the CPU usage by FS (<25%) until 150 concurrent calls when it starts to climb rapidly and at 175 (~50%). One good thing is that memory usage is reasonable with 90MB for 150 calls. My questions are; Q1 - Is my test reasonable ? Q2 - Windows XP is not that happy with large numbers of thread per app. Is there a way to configure FS to use fewer threads; instead of using 1/call can it be configured to use to 1/(n calls) ? Q3 - If there's not a way to configure FS to combine threads would it be reasonable to consider modifying the FS code to allow the combining of threads or would this be a mammoth task. Regards Richard Lamkin ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/bcd253b4/attachment.html From solko at gcdf.pl Thu Mar 26 12:46:32 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 26 Mar 2009 20:46:32 +0100 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> Message-ID: <49CBDB98.4090404@gcdf.pl> Richard Lamkin pisze: > Dear All, > > > > I am testing FS as a call queuing server. > > > > My test set up is as follows; > > Two Windows XP (SP3) Pc?s with 1.0.3 installed out of the box. [ > Before anyone says use Linux I need to use windows for a specific reason] > > > > I have a single gateway [fsb1500] on FS-1 configured to register with > FS-2 [extn 1500]. > > > > I then use the following CLI command to create a call from FS1[5900] to > FS2[5900] > > ?bgapi originate > sofia/gateway/fsb1500/5900 at richardl-5013-2.mettonigroup.com 5900? > > > > This happily connects FS1[5900] to FS2[5900] . Codec set to G711a. > > I have one hardphone on FS2 with a call connected to 5900 to allow me to > listen to MOH. > > > > I have been monitoring the system resources used by FS and have observed > that for each new call a new thread is created. > > I have pushed well over 200 calls from FS-1 into FS2. > > All goes well with the CPU usage by FS (<25%) until 150 concurrent calls > when it starts to climb rapidly and at 175 (~50%). > > One good thing is that memory usage is reasonable with 90MB for 150 calls. > > > > My questions are; > > Q1 ? Is my test reasonable ? > Memory usage looks good, I noticed on my laptop with Linux with Pentium M 1,6GHz that having something like 200 channels connected makes very big CPU usage, but what to expect if every channel was connected to one channel. > > > Q2 - Windows XP is not that happy with large numbers of thread per app. > Is there a way to configure FS to use fewer threads; instead of using > 1/call can it be configured to use to 1/(n calls) ? > Anthony here describes that there is one thread for channel, so don't expect it will change. > > > Q3 ? If there?s not a way to configure FS to combine threads would it > be reasonable to consider modifying the FS code to allow the combining > of threads or would this be a mammoth task. > The same as above. I don't know XP so well, but if it only has problems with many threads for one application then consider running multiple FS instances on one machine. > > > Regards > > Szymon Olko > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Mar 26 13:20:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 15:20:31 -0500 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> Message-ID: <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> The core philosophy of the entire design is based on a single thread per channel. There is no way to avoid this. What are the specs of your machines? DId you try using the 2003 server instead of XP (a home version meant for single users) We have little feedback on performance on win32 (good or bad) but we do have many people using it. Did you do any more extensive profiling to see what is using the most cpu (look for process explorer) 2009/3/26 Richard Lamkin > Dear All, > > > > I am testing FS as a call queuing server. > > > > My test set up is as follows; > > Two Windows XP (SP3) Pc?s with 1.0.3 installed out of the box. [ Before > anyone says use Linux I need to use windows for a specific reason] > > > > I have a single gateway [fsb1500] on FS-1 configured to register with FS-2 > [extn 1500]. > > > > I then use the following CLI command to create a call from FS1[5900] to > FS2[5900] > > ?bgapi originate sofia/gateway/fsb1500/ > 5900 at richardl-5013-2.mettonigroup.com 5900? > > > > This happily connects FS1[5900] to FS2[5900] . Codec set to G711a. > > I have one hardphone on FS2 with a call connected to 5900 to allow me to > listen to MOH. > > > > I have been monitoring the system resources used by FS and have observed > that for each new call a new thread is created. > > I have pushed well over 200 calls from FS-1 into FS2. > > All goes well with the CPU usage by FS (<25%) until 150 concurrent calls > when it starts to climb rapidly and at 175 (~50%). > > One good thing is that memory usage is reasonable with 90MB for 150 calls. > > > > My questions are; > > Q1 ? Is my test reasonable ? > > > > Q2 - Windows XP is not that happy with large numbers of thread per app. Is > there a way to configure FS to use fewer threads; instead of using 1/call > can it be configured to use to 1/(n calls) ? > > > > Q3 ? If there?s not a way to configure FS to combine threads would it be > reasonable to consider modifying the FS code to allow the combining of > threads or would this be a mammoth task. > > > > Regards > > > > Richard Lamkin > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/0104b63f/attachment.html From Prometheus001 at gmx.net Thu Mar 26 13:50:21 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 26 Mar 2009 21:50:21 +0100 Subject: [Freeswitch-users] Originate and Conference Message-ID: <49CBEA8D.4050901@gmx.net> Hello, when I originate a call via event socket and transfer it to a conference, it doesn't ask for a PIN. Example: originate sofia/default/222331 &conference(222500) The same happens when I originate a call and transfer it to the 222500 destination, which then transfers the call to the conference. originate sofia/default/222331 &transfer(222500) If I manually dial from the phone (222331) to the conference it correctly asks for the PIN. Anybody has a clue why this happens and how to enable the PIN while originating? Best regards Peter From msc at freeswitch.org Thu Mar 26 14:32:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Mar 2009 14:32:21 -0700 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <49CBEA8D.4050901@gmx.net> References: <49CBEA8D.4050901@gmx.net> Message-ID: <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> On Thu, Mar 26, 2009 at 1:50 PM, Peter P GMX wrote: > Hello, > > when I originate a call via event socket and transfer it to a > conference, it doesn't ask for a PIN. > Example: > originate sofia/default/222331 &conference(222500) > The same happens when I originate a call and transfer it to the 222500 > destination, which then transfers the call to the conference. > originate sofia/default/222331 &transfer(222500) > > If I manually dial from the phone (222331) to the conference it > correctly asks for the PIN. > > Anybody has a clue why this happens and how to enable the PIN while > originating? Is 222500 part of your dialplan? If so try this: originate sofia/default/222331 222500 Let us know if that works. -MC From peter at cindyandpeter.com Thu Mar 26 14:40:16 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Thu, 26 Mar 2009 17:40:16 -0400 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> Message-ID: <038c01c9ae5b$743cd310$5cb67930$@com> Anthony, No argument on the core design - this is what differentiates FS from * and what makes things solid. But, on Windows at least, the "one-thread-per-client" concept normally only scales so far (Server normally being a bit better than XP). Not sure about their availability on other platforms, but have you looked at fibers? They seem to be a lighter weight alternative with most of the benefits. Peter From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, March 26, 2009 4:21 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Load testing and thread use The core philosophy of the entire design is based on a single thread per channel. There is no way to avoid this. What are the specs of your machines? DId you try using the 2003 server instead of XP (a home version meant for single users) We have little feedback on performance on win32 (good or bad) but we do have many people using it. Did you do any more extensive profiling to see what is using the most cpu (look for process explorer) 2009/3/26 Richard Lamkin Dear All, I am testing FS as a call queuing server. My test set up is as follows; Two Windows XP (SP3) Pc's with 1.0.3 installed out of the box. [ Before anyone says use Linux I need to use windows for a specific reason] I have a single gateway [fsb1500] on FS-1 configured to register with FS-2 [extn 1500]. I then use the following CLI command to create a call from FS1[5900] to FS2[5900] "bgapi originate sofia/gateway/fsb1500/5900 at richardl-5013-2.mettonigroup.com 5900" This happily connects FS1[5900] to FS2[5900] . Codec set to G711a. I have one hardphone on FS2 with a call connected to 5900 to allow me to listen to MOH. I have been monitoring the system resources used by FS and have observed that for each new call a new thread is created. I have pushed well over 200 calls from FS-1 into FS2. All goes well with the CPU usage by FS (<25%) until 150 concurrent calls when it starts to climb rapidly and at 175 (~50%). One good thing is that memory usage is reasonable with 90MB for 150 calls. My questions are; Q1 - Is my test reasonable ? Q2 - Windows XP is not that happy with large numbers of thread per app. Is there a way to configure FS to use fewer threads; instead of using 1/call can it be configured to use to 1/(n calls) ? Q3 - If there's not a way to configure FS to combine threads would it be reasonable to consider modifying the FS code to allow the combining of threads or would this be a mammoth task. Regards Richard Lamkin ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/879aa827/attachment.html From brian at freeswitch.org Thu Mar 26 15:02:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 17:02:16 -0500 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <038c01c9ae5b$743cd310$5cb67930$@com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> <038c01c9ae5b$743cd310$5cb67930$@com> Message-ID: <526F1BEB-EC70-4234-976F-7F8777A89065@freeswitch.org> Is this windows specific? /b On Mar 26, 2009, at 4:40 PM, Peter J. Zandvoort wrote: > have you looked at fibers? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/b62ce3ef/attachment.html From brian at freeswitch.org Thu Mar 26 15:05:27 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 17:05:27 -0500 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <038c01c9ae5b$743cd310$5cb67930$@com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> <038c01c9ae5b$743cd310$5cb67930$@com> Message-ID: <7EFF1BA2-BA80-4454-9E77-A90F4452E987@freeswitch.org> OH so this is like coroutines. /b On Mar 26, 2009, at 4:40 PM, Peter J. Zandvoort wrote: > have you looked at fibers? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/73d3212b/attachment.html From grevenx at me.com Thu Mar 26 15:12:08 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Thu, 26 Mar 2009 23:12:08 +0100 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <7EFF1BA2-BA80-4454-9E77-A90F4452E987@freeswitch.org> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> <038c01c9ae5b$743cd310$5cb67930$@com> <7EFF1BA2-BA80-4454-9E77-A90F4452E987@freeswitch.org> Message-ID: <6E8BEF52-E107-45E9-84D7-D3F182203C2B@me.com> http://en.wikipedia.org/wiki/Fiber_(computer_science) Even Andr? Fiskvik On 26. mars. 2009, at 23.05, Brian West wrote: > OH so this is like coroutines. > > /b > > On Mar 26, 2009, at 4:40 PM, Peter J. Zandvoort wrote: > >> have you looked at fibers? > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/c64318dd/attachment-0001.html From telles-listas at devel-it.com.br Thu Mar 26 15:25:17 2009 From: telles-listas at devel-it.com.br (Rodrigo P. Telles) Date: Thu, 26 Mar 2009 19:25:17 -0300 Subject: [Freeswitch-users] Action and Anti-Action In-Reply-To: References: <49CB90D5.4090901@devel-it.com.br> Message-ID: <49CC00CD.6090703@devel-it.com.br> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/6c6ccdff/attachment.html From peter at cindyandpeter.com Thu Mar 26 15:33:29 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Thu, 26 Mar 2009 18:33:29 -0400 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <6E8BEF52-E107-45E9-84D7-D3F182203C2B@me.com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> <038c01c9ae5b$743cd310$5cb67930$@com> <7EFF1BA2-BA80-4454-9E77-A90F4452E987@freeswitch.org> <6E8BEF52-E107-45E9-84D7-D3F182203C2B@me.com> Message-ID: <03ad01c9ae62$e37a2230$aa6e6690$@com> Looks like it! I must admit that I have very little experience with it . it just looked like an additional toy to add to the bag of tricks. I?ve seen it used on mail servers and I think SQL Server now has the option. Both sound like things that would have long running client connections for which you want the simplicity of dedicated threads but not the overhead. For Windows: http://msdn.microsoft.com/en-us/library/ms682115(VS.85).aspx From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Even Andr? Fiskvik Sent: Thursday, March 26, 2009 6:12 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Load testing and thread use http://en.wikipedia.org/wiki/Fiber_(computer_science) Even Andr? Fiskvik On 26. mars. 2009, at 23.05, Brian West wrote: OH so this is like coroutines. /b On Mar 26, 2009, at 4:40 PM, Peter J. Zandvoort wrote: have you looked at fibers? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/bc00b04a/attachment.html From Prometheus001 at gmx.net Thu Mar 26 16:09:59 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Mar 2009 00:09:59 +0100 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> Message-ID: <49CC0B47.6000508@gmx.net> Hello Michael, I tried this, but received the same behaviour. It does not ask for the defined PIN. Best regards Peter Michael Collins schrieb: > On Thu, Mar 26, 2009 at 1:50 PM, Peter P GMX wrote: > >> Hello, >> >> when I originate a call via event socket and transfer it to a >> conference, it doesn't ask for a PIN. >> Example: >> originate sofia/default/222331 &conference(222500) >> The same happens when I originate a call and transfer it to the 222500 >> destination, which then transfers the call to the conference. >> originate sofia/default/222331 &transfer(222500) >> >> If I manually dial from the phone (222331) to the conference it >> correctly asks for the PIN. >> >> Anybody has a clue why this happens and how to enable the PIN while >> originating? >> > > Is 222500 part of your dialplan? If so try this: > > originate sofia/default/222331 222500 > > Let us know if that works. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu Mar 26 16:58:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Mar 2009 16:58:20 -0700 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <49CC0B47.6000508@gmx.net> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> Message-ID: <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> On Thu, Mar 26, 2009 at 4:09 PM, Peter P GMX wrote: > Hello Michael, > > I tried this, but received the same behaviour. It does not ask for the > defined PIN. Just curious - where do you define the PIN for this conference? -MC From asannucci at gmail.com Thu Mar 26 19:20:51 2009 From: asannucci at gmail.com (Andrea) Date: Thu, 26 Mar 2009 21:20:51 -0500 Subject: [Freeswitch-users] Error Compiling iksemel... References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com><1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> Message-ID: <1B117EFC18604C6E82663FDAEB3342A8@quos> try to install gnutls-devel (name on centos) before to compile. From kawarod at laposte.net Fri Mar 27 00:40:11 2009 From: kawarod at laposte.net (rod) Date: Fri, 27 Mar 2009 11:40:11 +0400 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 Message-ID: <49CC82DB.1000900@laposte.net> Hi, I did some tests with FS to transcode SIP INFO to RFC2833 (and vice versa) and it's working fine when FS stays in the media path with default configuration. But my setup is the following: - Core network requires SIP INFO - Peerings require RFC2833 all would be fine with FS if my SIP Peers were not enforcing G729 (discarding G711) so that I have to use the directive in my dialplan cause FS can't deal with G729 except in pass-through. It's sad, but G729 is still a reality in Telco World. So do you think there could be a way to deal with DTMF even if not analyzing RTP for transcoding. My commercial SBC is doing this, but it sucks and that's the last step before final migration to FS. regards, rod From jason at jasonjgw.net Fri Mar 27 00:58:20 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 27 Mar 2009 18:58:20 +1100 Subject: [Freeswitch-users] Multiple calls with PortAudio In-Reply-To: <20090326043931.GA6652@jdc.jasonjgw.net> References: <20090326043931.GA6652@jdc.jasonjgw.net> Message-ID: <20090327075820.GA15756@jdc.jasonjgw.net> While I was trying to obtain more detailed logs of my portaudio problems, FreeSWITCH crashed, leaving a core file. The backtraces are here: http://pastebin.freeswitch.org/7998 As far as I can remember, at the time of the segfault, one channel was trying to connect and not succeeding; I had just issued a pa hangup command on it and then a pa call to try connecting again. Since my memory of exactly what was happening isn't as reliable as it should be, the value of the backtraces may be diminished. As to the portaudio problem, with rev. 12701 (Debian Sid, kernel 2.6.29, x86_64 architecture), the situation appears to be that the second and subsequent concurrent portaudio calls sometimes wait for a long time after issuing a log message such as the following: [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel sofia/internal/1000 at 192.168.0.2:5070 [d6e56642-1a9b-11de-b23e-c5a9450df57d] These calls do not always complete successfully, but I'm still trying to collect more precise details of when and why they fail. With apologies for being unable to use Jira, if anything valuable appears in the backtraces, you are welcome to let me know via the list or by e-mail. I have previously seen crashes while working with multiple portaudio calls, but I don't yet have a reliable means to reproduce them. If the backtraces are revealing, that's good, but if not, that's fine too and I'll collect better particulars next time. From Prometheus001 at gmx.net Fri Mar 27 01:27:42 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Mar 2009 09:27:42 +0100 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> Message-ID: <49CC8DFE.3050104@gmx.net> It's defined via XML-Curl, and manual dialling and transfering do trigger the same xml-curl request. This means that this conference number is not defined in the any xml conf file. If I transfer a call (without PIN) and then manually dial with another phone into this conf with PIN, both calls are in the same conference. I have SVN rev 12796. Best regards Peter Michael Collins schrieb: > On Thu, Mar 26, 2009 at 4:09 PM, Peter P GMX wrote: > >> Hello Michael, >> >> I tried this, but received the same behaviour. It does not ask for the >> defined PIN. >> > > Just curious - where do you define the PIN for this conference? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ttroy50 at gmail.com Fri Mar 27 02:55:04 2009 From: ttroy50 at gmail.com (Thomas Troy) Date: Fri, 27 Mar 2009 09:55:04 +0000 Subject: [Freeswitch-users] sipp emulating a registered end point In-Reply-To: References: Message-ID: When I need to do something like this, what I do is set sipp2 to have 2 scripts both using the same IP and port. One sends the register and deals with authentication \ OK. The other is then run after this to wait and receive the incoming call. So you would run them like sipp2_register - Performs registration and ends sipp2_receiveCall - Waits for incoming call, while listening on same IP \ Port as sipp2_register sipp1_makeCall - Makes outgoing call A sample register scenario is From: TestUser1 ;tag=[pid][call_number] Call-ID: [call_id] CSeq: 1 REGISTER Contact: Expires: 240 User-Agent: SIPp Content-Length: 0 ]]> From: TestUser1 ;tag=[pid][call_number] Call-ID: [call_id] CSeq: 2 REGISTER Contact: Expires: 240 User-Agent: SIPp [authentication username=user password=pass] Content-Length: 0 ]]> 2009/3/26 Jonas Gauffin > Hello > I want to achive this: Sipp1 -> FS -> Sipp2 > > Sipp1 emulates a inbound calls (easy to achive) > Sipp2 should emulate a registered user (i.e. register with FS and then just > wait for calls and hangup when sipp1 hangsup) > > How do I configure sipp as "Sipp2"? > > Thanks, > Jonas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/4a25a7c5/attachment.html From mattdfong at gmail.com Fri Mar 27 04:48:45 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 27 Mar 2009 18:48:45 +0700 Subject: [Freeswitch-users] rubymod - ESL compile error Message-ID: <4256bf830903270448g5edc1a22sba2a91ea85fdbc1b@mail.gmail.com> I'm trying to get rubymod, working to experiment with it, but I'm getting the following error when I try to make on my Ubuntu system. root at ubuntu:/usr/src/freeswitch/libs/esl# make rubymod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C ruby make[1]: Entering directory `/usr/src/freeswitch/libs/esl/ruby' g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L. /usr/bin/ld: cannot find -lruby collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/ruby' make: *** [rubymod] Error 2 I'm currently using event sockets with a fully ruby implementation, but it's sort of slow at reading sockets. If I can get it working, it will be interesting seeing if I can improve performance. Does rubymod support events the same way the perlmod does? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/6fa833b2/attachment.html From mattdfong at gmail.com Fri Mar 27 05:12:50 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 27 Mar 2009 19:12:50 +0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based Message-ID: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> I've been playing around with using freeswitch.EventConsumer in a lua process that starts-up when FS boots, and stays in the background. I've setup the example on the wiki, but the example uses session:execute("sleep",1000), and essentially loops every second until an event is fired. I'm wondering if there is a more event-driven way to accomplish this? I tried asking for help in #lua, but they said the project (FS) needed to implement event-driven programming for this to work. To me, it seems sort of silly to implement freeswitch.EventConsumer without a way for it to be executed event-wise Is using lua ESL the only option? There isn't any lua example scripts in libs/esl/lua to demonstrate how to handle events. if mod_lua can't handle events, can the mod_javascript utilize it? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/024a368a/attachment.html From andy at fabulous4.co.uk Fri Mar 27 05:34:46 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Fri, 27 Mar 2009 12:34:46 -0000 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <2365564C-92A8-483E-9FCB-34EBE71EC256@avgs.ca> Message-ID: Thanks for your help folks, the ping parameter seems to have resolved the gateway connection issue but I now seem to be having a related issue with calls being cut off after a number of seconds. The freeswitch logs show a normal call clearing. I am indeed behind a NAT firewall which I'm assuming is the main issue. do you have any further tips to make this more stable and prevent the call cut off? Many thanks Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: 18 March 2009 14:46 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Losing Gateway registration if you are behind NAT it is possible that your router "forgot" the mapping betweeen FS and your provider, try adding to your gateway. Math On 18-Mar-09, at 10:07 AM, Brian West wrote: Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/b5877b66/attachment.html From brian at freeswitch.org Fri Mar 27 06:50:30 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 08:50:30 -0500 Subject: [Freeswitch-users] Multiple calls with PortAudio In-Reply-To: <20090327075820.GA15756@jdc.jasonjgw.net> References: <20090326043931.GA6652@jdc.jasonjgw.net> <20090327075820.GA15756@jdc.jasonjgw.net> Message-ID: <9D237452-A946-492C-ABC2-FEDDF453E9AF@freeswitch.org> Please direct the report to http://jira.freeswitch.org /b On Mar 27, 2009, at 2:58 AM, Jason White wrote: > While I was trying to obtain more detailed logs of my portaudio > problems, > FreeSWITCH crashed, leaving a core file. > > The backtraces are here: > http://pastebin.freeswitch.org/7998 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/1fd06fd5/attachment-0001.html From brian at freeswitch.org Fri Mar 27 06:49:09 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 08:49:09 -0500 Subject: [Freeswitch-users] rubymod - ESL compile error In-Reply-To: <4256bf830903270448g5edc1a22sba2a91ea85fdbc1b@mail.gmail.com> References: <4256bf830903270448g5edc1a22sba2a91ea85fdbc1b@mail.gmail.com> Message-ID: <763DC25A-FECF-4E40-9089-4C8CCC5B8943@freeswitch.org> http://jira.freeswitch.org/browse/ESL-7 I think this might apply to you. /b On Mar 27, 2009, at 6:48 AM, Matthew Fong wrote: > /usr/bin/ld: cannot find -lruby > collect2: ld returned 1 exit status Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/4ef56170/attachment.html From jonas.gauffin at gmail.com Fri Mar 27 07:05:13 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Fri, 27 Mar 2009 15:05:13 +0100 Subject: [Freeswitch-users] sipp emulating a registered end point In-Reply-To: References: Message-ID: thanks! 2009/3/27 Thomas Troy > When I need to do something like this, what I do is set sipp2 to have 2 > scripts both using the same IP and port. > > One sends the register and deals with authentication \ OK. > The other is then run after this to wait and receive the incoming call. > > So you would run them like > > sipp2_register - Performs registration and ends > sipp2_receiveCall - Waits for incoming call, while listening > on same IP \ Port as sipp2_register > sipp1_makeCall - Makes outgoing call > > > A sample register scenario is > > > > > > REGISTER sip:[remote_ip]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] > [local_ip]:[local_port];branch=[branch]1;rport > Max-Forwards: 70 > To: TestUser1 > From: TestUser1 [remote_ip]:[remote_port]>;tag=[pid][call_number] > Call-ID: [call_id] > CSeq: 1 REGISTER > Contact: > Expires: 240 > User-Agent: SIPp > Content-Length: 0 > > ]]> > > > > > > > > REGISTER sip:[remote_ip]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] > [local_ip]:[local_port];branch=[branch]2;rport > Max-Forwards: 70 > To: TestUser1 > From: TestUser1 [remote_ip]:[remote_port]>;tag=[pid][call_number] > Call-ID: [call_id] > CSeq: 2 REGISTER > Contact: > Expires: 240 > User-Agent: SIPp > [authentication username=user password=pass] > Content-Length: 0 > > ]]> > > > > > > > > > > > > > > > 2009/3/26 Jonas Gauffin > >> Hello >> I want to achive this: Sipp1 -> FS -> Sipp2 >> >> Sipp1 emulates a inbound calls (easy to achive) >> Sipp2 should emulate a registered user (i.e. register with FS and then >> just wait for calls and hangup when sipp1 hangsup) >> >> How do I configure sipp as "Sipp2"? >> >> Thanks, >> Jonas >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/03dc6d51/attachment.html From brian at freeswitch.org Fri Mar 27 07:18:27 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 09:18:27 -0500 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <49CC82DB.1000900@laposte.net> References: <49CC82DB.1000900@laposte.net> Message-ID: <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> On Mar 27, 2009, at 2:40 AM, rod wrote: > Hi, > > I did some tests with FS to transcode SIP INFO to RFC2833 (and vice > versa) and it's working fine when FS stays in the media path with > default configuration. > > But my setup is the following: > - Core network requires SIP INFO > - Peerings require RFC2833 > > all would be fine with FS if my SIP Peers were not enforcing G729 > (discarding G711) so that I have to use the directive application="set" data="proxy_media=true"/> in my dialplan cause FS > can't deal with G729 except in pass-through. Can't use proxy media in this case. (I highly recommend you not use Proxy Media mode) > > It's sad, but G729 is still a reality in Telco World. Coming soon! > > So do you think there could be a way to deal with DTMF even if not > analyzing RTP for transcoding. My commercial SBC is doing this, but it > sucks and that's the last step before final migration to FS. > > regards, > rod Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/09438dc8/attachment.html From anthony.minessale at gmail.com Fri Mar 27 07:18:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 09:18:10 -0500 Subject: [Freeswitch-users] Multiple calls with PortAudio In-Reply-To: <20090327075820.GA15756@jdc.jasonjgw.net> References: <20090326043931.GA6652@jdc.jasonjgw.net> <20090327075820.GA15756@jdc.jasonjgw.net> Message-ID: <191c3a030903270718k4d5dbdfvd19e57b5bad381cf@mail.gmail.com> Are you updating with "make current" each time? On Fri, Mar 27, 2009 at 2:58 AM, Jason White wrote: > While I was trying to obtain more detailed logs of my portaudio problems, > FreeSWITCH crashed, leaving a core file. > > The backtraces are here: > http://pastebin.freeswitch.org/7998 > > As far as I can remember, at the time of the segfault, one channel was > trying > to connect and not succeeding; I had just issued a pa hangup command on it > and > then a pa call to try connecting again. Since my memory of exactly what was > happening isn't as reliable as it should be, the value of the backtraces > may > be diminished. > > As to the portaudio problem, with rev. 12701 (Debian Sid, kernel 2.6.29, > x86_64 architecture), the situation appears to be that the second and > subsequent concurrent portaudio calls sometimes wait for a long time after > issuing a log message such as the following: > > [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel > sofia/internal/1000 at 192.168.0.2:5070[d6e56642-1a9b-11de-b23e-c5a9450df57d] > > These calls do not always complete successfully, but I'm still trying to > collect more precise details of when and why they fail. > > With apologies for being unable to use Jira, if anything valuable appears > in > the backtraces, you are welcome to let me know via the list or by e-mail. > > I have previously seen crashes while working with multiple portaudio calls, > but I don't yet have a reliable means to reproduce them. > > If the backtraces are revealing, that's good, but if not, that's fine too > and > I'll collect better particulars next time. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/52a6ca02/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 27 07:52:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 09:52:48 -0500 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> Message-ID: <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> Sort of silly?, I am not sure what you are talking about. I t's called *event*Consumer right? what do you mean by event based? There is no need to create a session? con = freeswitch.EventConsumer("all"); now you have a consumer obj every time you call con:pop() with no arg you will either get an event or nil when there are no events to consume. every time you call con:pop(1) the consumer object will block until there is an event. So you use the first way in conjunction with some other lock to do async or the 2nd way you do a dedicated blocking loop. I don't know what you said in #lua but, umm duhhhh I think we have an event driven programming under control..... We have a dedicated eventing engine in the core with scaling backend dispatcher threads that can handle hundereds of thousand of events at a time. There is also Event Socket (the word *event* again) that can connect to tcp and listen for *events* you can also write your code in C with the trivial module API that allows you to bind to an event internally and pretty much do whatever you want. 2009/3/27 Matthew Fong > I've been playing around with using freeswitch.EventConsumer in a lua > process that starts-up when FS boots, and stays in the background. I've > setup the example on the wiki, but the example uses > session:execute("sleep",1000), and essentially loops every second until an > event is fired. I'm wondering if there is a more event-driven way to > accomplish this? > I tried asking for help in #lua, but they said the project (FS) needed to > implement event-driven programming for this to work. To me, it seems sort of > silly to implement freeswitch.EventConsumer without a way for it to be > executed event-wise > > Is using lua ESL the only option? There isn't any lua example scripts in > libs/esl/lua to demonstrate how to handle events. > > if mod_lua can't handle events, can the mod_javascript utilize it? Thanks. > > --matt > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/451757a5/attachment.html From kawarod at laposte.net Fri Mar 27 08:07:51 2009 From: kawarod at laposte.net (rod) Date: Fri, 27 Mar 2009 19:07:51 +0400 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> References: <49CC82DB.1000900@laposte.net> <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> Message-ID: <49CCEBC7.1070807@laposte.net> Hi Brian, don't understand very well your advice: --> Can't use proxy media in this case. (I highly recommend you not use Proxy Media mode) If i want to hide my topology network and deal with G729, I must use proxy media ? Why is Proxy media mode not recommended ?? regards. rod Brian West wrote: > > On Mar 27, 2009, at 2:40 AM, rod wrote: > >> Hi, >> >> I did some tests with FS to transcode SIP INFO to RFC2833 (and vice >> versa) and it's working fine when FS stays in the media path with >> default configuration. >> >> But my setup is the following: >> - Core network requires SIP INFO >> - Peerings require RFC2833 >> >> all would be fine with FS if my SIP Peers were not enforcing G729 >> (discarding G711) so that I have to use the directive > application="set" data="proxy_media=true"/> in my dialplan cause FS >> can't deal with G729 except in pass-through. > > Can't use proxy media in this case. (I highly recommend you not use > Proxy Media mode) > >> >> It's sad, but G729 is still a reality in Telco World. > > Coming soon! > >> >> So do you think there could be a way to deal with DTMF even if not >> analyzing RTP for transcoding. My commercial SBC is doing this, but it >> sucks and that's the last step before final migration to FS. >> >> regards, >> rod > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From helmut.kuper at ewetel.de Fri Mar 27 08:09:09 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 27 Mar 2009 16:09:09 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CB9E0C.4030300@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> <49CB9E0C.4030300@ewetel.de> Message-ID: <49CCEC15.8010500@ewetel.de> Hello, today I killed that special thread via "kill -9" a simple kill didn't helped. Unfortunately this led to a normal shutdown of FS although I killed not the parent process. :( After restart of FS the server has a normal load again. regards and a nice weekend Helmut From anthony.minessale at gmail.com Fri Mar 27 08:13:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 10:13:59 -0500 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <49CCEBC7.1070807@laposte.net> References: <49CC82DB.1000900@laposte.net> <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> <49CCEBC7.1070807@laposte.net> Message-ID: <191c3a030903270813x1319825bva59b09b789f79855@mail.gmail.com> if you enable mod_g729 you can use freeswitch normally with that g729 codec as long as no transcoding is enabled (same passthru concept as proxy_media_mode) On Fri, Mar 27, 2009 at 10:07 AM, rod wrote: > Hi Brian, > > don't understand very well your advice: > --> Can't use proxy media in this case. (I highly recommend you not use > Proxy Media mode) > > If i want to hide my topology network and deal with G729, I must use > proxy media ? > Why is Proxy media mode not recommended ?? > > regards. > rod > > > > Brian West wrote: > > > > On Mar 27, 2009, at 2:40 AM, rod wrote: > > > >> Hi, > >> > >> I did some tests with FS to transcode SIP INFO to RFC2833 (and vice > >> versa) and it's working fine when FS stays in the media path with > >> default configuration. > >> > >> But my setup is the following: > >> - Core network requires SIP INFO > >> - Peerings require RFC2833 > >> > >> all would be fine with FS if my SIP Peers were not enforcing G729 > >> (discarding G711) so that I have to use the directive >> application="set" data="proxy_media=true"/> in my dialplan cause FS > >> can't deal with G729 except in pass-through. > > > > Can't use proxy media in this case. (I highly recommend you not use > > Proxy Media mode) > > > >> > >> It's sad, but G729 is still a reality in Telco World. > > > > Coming soon! > > > >> > >> So do you think there could be a way to deal with DTMF even if not > >> analyzing RTP for transcoding. My commercial SBC is doing this, but it > >> sucks and that's the last step before final migration to FS. > >> > >> regards, > >> rod > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/8c4491f2/attachment.html From jht at lj.net Fri Mar 27 03:11:45 2009 From: jht at lj.net (James H Thompson) Date: Fri, 27 Mar 2009 00:11:45 -1000 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls Message-ID: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> I need to generate calls with Invite URIs in this format: INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 SIP/2.0 Is there an easy way to do this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/42a5026e/attachment-0001.html From fdelawarde at wirelessmundi.com Fri Mar 27 05:05:04 2009 From: fdelawarde at wirelessmundi.com (Francois Delawarde) Date: Fri, 27 Mar 2009 13:05:04 +0100 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues Message-ID: <1238155504.4364.222.camel@localhost.localdomain> Hello and welcome me into FreeSWITCH's world! <= sorry that was rude I am (hoping to say "I was" soon) a heavy user of Asterisk's call queues for small call centers with sometimes empty queues and all agents idle for a few seconds. FreeSWITCH's mod_fifo algorithm is apparently quite different than Asterisk's app_queue. Instead of choosing an agent for a each call once it gets to the bottom of the queue given a specific strategy, FreeSWITCH does the inverse and finds a call once an agent is free given a strategy (the call that has waited longer from all the agent's queues, or the call in the queue that currently has more calls waiting). Am I right? If the above deduction is correct, while it seems a MUCH better choice for heavier call centers that always have calls in their queues ("in queue" calls are not delayed by the processing of the call at the end of the queue), I have a few doubts for what would happen in small call centers when those queues sometimes get empty and several agents "fight" for the incoming calls. My questions are following: - If for example 4 agents are "connected" (fifo out) to an empty queue, what happens when a call arrives? Do the 4 agents ring? If not, how do we know which agent get the call? - Is there an [easy] way (with some javascript or similar) to "emulate" Asterisk's distribution strategies to agents (by amount of time without calls, total number of answered calls, round robing, ...) in this cases? A couple of other newbie questions that has nothing to do with the above: - Is there a way to execute some PHP scripts for each call that would do the bridging or call applications (like Asterisk's AGI)? - What is the recommended language for features, speed, and programming ease (not a priority) for this kind of scripts (C? LUA?, Javascript?, ..)? Thanks in advance, Fran?ois. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/98fbb298/attachment-0001.html From mike at jerris.com Fri Mar 27 08:17:56 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Mar 2009 11:17:56 -0400 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: References: Message-ID: <6E440475-070D-4C39-B5A3-AC91781B4555@jerris.com> Look closer at the logs or sip trace, this sounds like a failed session timer to me. Mike On Mar 27, 2009, at 8:34 AM, Andy Ayers wrote: > Thanks for your help folks, the ping parameter seems to have > resolved the gateway connection issue but I now seem to be having a > related issue with calls being cut off after a number of seconds. > The freeswitch logs show a normal call clearing. I am indeed behind > a NAT firewall which I'm assuming is the main issue. do you have any > further tips to make this more stable and prevent the call cut off? > > Many thanks > Andy > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Mathieu Rene > Sent: 18 March 2009 14:46 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Losing Gateway registration > > if you are behind NAT it is possible that your router "forgot" the > mapping betweeen FS and your provider, try adding name="ping" value="30" /> to your gateway. > > Math > > On 18-Mar-09, at 10:07 AM, Brian West wrote: > >> Upgrade to 1.03 or SVN Trunk >> >> /b >> >> On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: >> >>> Hi, >>> >>> I've recently ugrade to version 1.02 of freeswitch and am having >>> some problems with my gateway registrations. The gateway >>> successfully registers with my voip provider when freeswitch first >>> starts but if left running it seems to loose it's connection to my >>> voip provider. I can get it to reconnect with a sofia restart. I'm >>> using the same provider and user account as with the old version >>> of the software. Can you suggest any reaosn why this may be >>> happening and how I can prevent it? >>> >>> Many thanks >>> Andy >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/09b10649/attachment.html From brian at freeswitch.org Fri Mar 27 08:23:17 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 10:23:17 -0500 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> Message-ID: <09BC8800-D604-40C3-B682-43DFF20DF3B0@freeswitch.org> Params are not added before the @ sign that I'm pretty sure of. Results in INVITE sip:blah at blah;this=kewl;that=iskewl SIP/2.0 You also have sip_invite_to_params, sip_invite_from_params, sip_invite_contact_params /b On Mar 27, 2009, at 5:11 AM, James H Thompson wrote: > I need to generate calls with Invite URIs in this format: > > INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 > SIP/2.0 > > Is there an easy way to do this? > > Thanks. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/442a3579/attachment.html From mike at jerris.com Fri Mar 27 08:32:37 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Mar 2009 11:32:37 -0400 Subject: [Freeswitch-users] rubymod - ESL compile error In-Reply-To: <763DC25A-FECF-4E40-9089-4C8CCC5B8943@freeswitch.org> References: <4256bf830903270448g5edc1a22sba2a91ea85fdbc1b@mail.gmail.com> <763DC25A-FECF-4E40-9089-4C8CCC5B8943@freeswitch.org> Message-ID: <28E9E7EB-7F31-4CEE-89EC-5348CC055E1D@jerris.com> fixed revision 12805. Mike On Mar 27, 2009, at 9:49 AM, Brian West wrote: > http://jira.freeswitch.org/browse/ESL-7 > > I think this might apply to you. > > /b > > On Mar 27, 2009, at 6:48 AM, Matthew Fong wrote: > >> /usr/bin/ld: cannot find -lruby >> collect2: ld returned 1 exit status > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/19328339/attachment.html From mike at jerris.com Fri Mar 27 08:41:46 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Mar 2009 11:41:46 -0400 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> Message-ID: You seem to be confusing your standards, those 2 specs are about tel: uri's not sip: uris. Sending a tel uri I am not sure we can do, where would we send it to? Mike On Mar 27, 2009, at 6:11 AM, James H Thompson wrote: > I need to generate calls with Invite URIs in this format: > > INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 > SIP/2.0 > > Is there an easy way to do this? > > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/d8698a1b/attachment.html From anthony.minessale at gmail.com Fri Mar 27 08:55:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 10:55:46 -0500 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues In-Reply-To: <1238155504.4364.222.camel@localhost.localdomain> References: <1238155504.4364.222.camel@localhost.localdomain> Message-ID: <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> 2009/3/27 Francois Delawarde > Hello and welcome me into FreeSWITCH's world! <= sorry that was rude > > I am (hoping to say "I was" soon) a heavy user of Asterisk's call queues > for small call centers with sometimes empty queues and all agents idle for a > few seconds. > > FreeSWITCH's mod_fifo algorithm is apparently quite different than > Asterisk's app_queue. Instead of choosing an agent for a each call once it > gets to the bottom of the queue given a specific strategy, FreeSWITCH does > the inverse and finds a call once an agent is free given a strategy (the > call that has waited longer from all the agent's queues, or the call in the > queue that currently has more calls waiting). Am I right? > > If the above deduction is correct, while it seems a MUCH better choice for > heavier call centers that always have calls in their queues ("in queue" > calls are not delayed by the processing of the call at the end of the > queue), I have a few doubts for what would happen in small call centers when > those queues sometimes get empty and several agents "fight" for the incoming > calls. My questions are following: > > - If for example 4 agents are "connected" (fifo out) to an empty queue, > what happens when a call arrives? Do the 4 agents ring? If not, how do we > know which agent get the call? > If you are using on-hook agents, it will place as many outbound calls as there are people waiting. If you are using off-hook agents it will just connect the first free agent. > - Is there an [easy] way (with some javascript or similar) to "emulate" > Asterisk's distribution strategies to agents (by amount of time without > calls, total number of answered calls, round robing, ...) in this cases? > Easiest way would be to write a patch in C to mod_fifo it'self or propose a bounty for features and see if you can get the change approved by the developers. > > A couple of other newbie questions that has nothing to do with the above: > - Is there a way to execute some PHP scripts for each call that would do > the bridging or call applications (like Asterisk's AGI)? > Your best bet would be to not try to do anything "like asterisk" FreeSWITCH is a paradigm shift from asterisk and you may defeat yourself trying to do anything the same way. That said, yes, look at Event Socket and ESL, (using asterisk terminology, it's a combination of AGI and manager). > - What is the recommended language for features, speed, and programming > ease (not a priority) for this kind of scripts (C? LUA?, Javascript?, ..)? > C > > Thanks in advance, > Fran?ois. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/09af9a02/attachment-0001.html From dujinfang at gmail.com Fri Mar 27 09:12:17 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 28 Mar 2009 00:12:17 +0800 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> Message-ID: <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: > We do not support ubuntu interpid, it has at least 3 known fatal > issues not experienced by all but nonetheless enough to make us > unwilling to support it. I use Ubuntu gutsy in production and interipid in test env. It works well. Can you briefly explain the 3 fatal issues Anthony? It will help me know potential risks. > > > It's "use at your own risk" or use the stable branch "hardy" for any > support. > > > > On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds > wrote: > Has there been any progress getting FreeSWITCH to build on Ubuntu > Intrepid without downgrading libtool? > I successfully built FS on intrepid. I simply done this by changing the apt-source to Hardy and installed libtool. Obviously I changed the apt-source back to intrepid after I installed libtool. And, another approach. Install libtool from source should be as easy as configure && make && make install. I done this on a new CentOS4 because the default yum install of libtool on CentOS4 is old than FS required. > Thanks! > > Sincerely, > Trevor Hammonds > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/a68ea737/attachment.html From kawarod at laposte.net Fri Mar 27 09:23:59 2009 From: kawarod at laposte.net (rod) Date: Fri, 27 Mar 2009 20:23:59 +0400 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <191c3a030903270813x1319825bva59b09b789f79855@mail.gmail.com> References: <49CC82DB.1000900@laposte.net> <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> <49CCEBC7.1070807@laposte.net> <191c3a030903270813x1319825bva59b09b789f79855@mail.gmail.com> Message-ID: <49CCFD9F.70600@laposte.net> Hello, I have this error when not enablig proxy_media: 2009-03-27 19:54:44 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-27 19:54:44 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! Sure there is an option to check. Any pointers. regards. Anthony Minessale wrote: > if you enable mod_g729 you can use freeswitch normally with that g729 > codec as long > as no transcoding is enabled (same passthru concept as proxy_media_mode) > > > On Fri, Mar 27, 2009 at 10:07 AM, rod > wrote: > > Hi Brian, > > don't understand very well your advice: > --> Can't use proxy media in this case. (I highly recommend you > not use > Proxy Media mode) > > If i want to hide my topology network and deal with G729, I must use > proxy media ? > Why is Proxy media mode not recommended ?? > > regards. > rod > > > > Brian West wrote: > > > > On Mar 27, 2009, at 2:40 AM, rod wrote: > > > >> Hi, > >> > >> I did some tests with FS to transcode SIP INFO to RFC2833 (and vice > >> versa) and it's working fine when FS stays in the media path with > >> default configuration. > >> > >> But my setup is the following: > >> - Core network requires SIP INFO > >> - Peerings require RFC2833 > >> > >> all would be fine with FS if my SIP Peers were not enforcing G729 > >> (discarding G711) so that I have to use the directive >> application="set" data="proxy_media=true"/> in my dialplan cause FS > >> can't deal with G729 except in pass-through. > > > > Can't use proxy media in this case. (I highly recommend you not use > > Proxy Media mode) > > > >> > >> It's sad, but G729 is still a reality in Telco World. > > > > Coming soon! > > > >> > >> So do you think there could be a way to deal with DTMF even if not > >> analyzing RTP for transcoding. My commercial SBC is doing this, > but it > >> sucks and that's the last step before final migration to FS. > >> > >> regards, > >> rod > > > > Brian West > > brian at freeswitch.org > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Fri Mar 27 09:25:06 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Mar 2009 17:25:06 +0100 Subject: [Freeswitch-users] condition matching on variables which have been set in the dialplan Message-ID: <49CCFDE2.4050704@gmx.net> I try to use speed dialling and masked numbers in a dialplan through xml-curl. For the XML I use templates which I fill with variables. The numbering plan is set up in a way that any number can be a speed dialling or masked number, so I cannot parse them via Regex in the XML part of the dialplan. E.g. * 12345 is a normal phone * 12346 is a speed dialling number => 0049xxxxxxxxxx * 12347 is a normal phone * 4 is a speed dialling number => 0049xxxxxxxxxx So I need to substitute a variable with the final number to be dialled. This final number then needs to be parsed in the dialplan to indentify how to handle it (bridge, conference, voicemail etc.) I have special reasons to do that, so please do not ask me why. So the dialplan is as following . . . . . . . . In the first condition I set the substituted final destination number. This is dynamically substituted in the template in my application via xml-curl dependend on which kind of number is dialled. In this case a German number is substituted. In the following conditions I would like to set the gateways. What is happening in the logs? * I dial e.g. "12346" for a speed dialling number * The first condition is parsed correctly, and the variables are set (Action set(destination_number=0049xxxxxxxxxxxx) * in the second condition "${variable_destination_number} is not set to the new value. It's still "12346".(I also tried conditions based on "${destination_number}" and "destination_number"). * In the logs the execution of "set" and "export" in fact is shown the whole conditions are parsed. Also the info application is outputted after all conditions are parsed. E.g. EXECUTE sofia/internal/10000 at sip.domain.de set(destination_number=0049xxxxxxxxxxxx) * the "info" app shows me that "variable_destination_number" is set to the right number, but it seems to be too late? Question: Are these lines not handled sequentially (I am using a quad core machine)? Any other idea how to solve this? Best regards Peter From brian at freeswitch.org Fri Mar 27 09:55:08 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 11:55:08 -0500 Subject: [Freeswitch-users] condition matching on variables which have been set in the dialplan In-Reply-To: <49CCFDE2.4050704@gmx.net> References: <49CCFDE2.4050704@gmx.net> Message-ID: <7C44CF00-5B22-4E19-A0A8-74C820E02A46@freeswitch.org> Remember the dialplan is NOT executed when its parsed so you can't set a var then condition on that exact var on the next line.. that var doesn't exist. /b On Mar 27, 2009, at 11:25 AM, Peter P GMX wrote: > > expression="^[0-9]\d[0,16}$" > continue="true">. > data="destination_number=0049xxxxxxxxx"/>. > data="destination_number=0049xxxxxxxxx"/>. > > > expression="^(00[1-9]\d{4,13})$"> > data="effective_caller_id_number=unknown"/>. > data="effective_caller_id_name=unknown"/>. > data="sofia/gateway/QSC_DE/$1 at sip.qsc.de"/>. > > > . > . > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/6ca27cc8/attachment.html From mike at jerris.com Fri Mar 27 10:31:09 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Mar 2009 13:31:09 -0400 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> Message-ID: <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> Another example of a fatal issue was the optimizer in gcc was breaking openzap code even with -O2. Mike On Mar 27, 2009, at 12:12 PM, dujinfang wrote: > > On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: >> We do not support ubuntu interpid, it has at least 3 known fatal >> issues not experienced by all but nonetheless enough to make us >> unwilling to support it. > > I use Ubuntu gutsy in production and interipid in test env. It > works well. Can you briefly explain the 3 fatal issues Anthony? It > will help me know potential risks. >> >> It's "use at your own risk" or use the stable branch "hardy" for >> any support. >> >> On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds > > wrote: >> Has there been any progress getting FreeSWITCH to build on Ubuntu >> Intrepid without downgrading libtool? >> > > I successfully built FS on intrepid. I simply done this by changing > the apt-source to Hardy and installed libtool. Obviously I changed > the apt-source back to intrepid after I installed libtool. > > And, another approach. Install libtool from source should be as easy > as configure && make && make install. I done this on a new CentOS4 > because the default yum install of libtool on CentOS4 is old than FS > required. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/a332d915/attachment-0001.html From msc at freeswitch.org Fri Mar 27 10:43:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 27 Mar 2009 10:43:13 -0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> Message-ID: <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> > con = freeswitch.EventConsumer("all"); > > now you have a consumer obj > > every time you call con:pop() with no arg you will either get an event or > nil when there are no events to consume. > every time you call con:pop(1) the consumer object will block until there is > an event. > > So you use the first way in conjunction with some other lock to do async or > the 2nd way you do a dedicated blocking loop. FYI, I added this information to the wiki page for freeswitch.EventConsumer. -MC From gkuri at ieee.org Fri Mar 27 11:19:57 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 27 Mar 2009 11:19:57 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> Message-ID: <49CD18CD.5000704@ieee.org> regarding gcc compiler optimizations, are they generally compatible with FS or should they be removed or does the configure strip them out? just curious, as I run Gentoo and use such optimizations as "-march=nocona -O2 -pipe -fomit-frame-pointer" not sure if they break things or I should be removing them before compiling FS? Gabe Michael Jerris wrote: > Another example of a fatal issue was the optimizer in gcc was breaking > openzap code even with -O2. > > Mike > > On Mar 27, 2009, at 12:12 PM, dujinfang wrote: > >> >> On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: >>> We do not support ubuntu interpid, it has at least 3 known fatal >>> issues not experienced by all but nonetheless enough to make us >>> unwilling to support it. >> >> I use Ubuntu gutsy in production and interipid in test env. It works >> well. Can you briefly explain the 3 fatal issues Anthony? It will help >> me know potential risks. >>> >>> It's "use at your own risk" or use the stable branch "hardy" for any >>> support. >>> >>> On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds >>> > wrote: >>> >>> Has there been any progress getting FreeSWITCH to build on Ubuntu >>> Intrepid without downgrading libtool? >>> >> >> I successfully built FS on intrepid. I simply done this by changing >> the apt-source to Hardy and installed libtool. Obviously I changed the >> apt-source back to intrepid after I installed libtool. >> >> And, another approach. Install libtool from source should be as easy >> as configure && make && make install. I done this on a new CentOS4 >> because the default yum install of libtool on CentOS4 is old than FS >> required. > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri Mar 27 11:30:01 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 13:30:01 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <49CD18CD.5000704@ieee.org> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> Message-ID: Usually if you don't know what they do... then you shouldn't use them! ;) /b On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote: > regarding gcc compiler optimizations, are they generally compatible > with > FS or should they be removed or does the configure strip them out? > just > curious, as I run Gentoo and use such optimizations as "-march=nocona > -O2 -pipe -fomit-frame-pointer" > > not sure if they break things or I should be removing them before > compiling FS? > > Gabe Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/11f858c1/attachment.html From anthony.minessale at gmail.com Fri Mar 27 11:55:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 13:55:27 -0500 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <49CCFD9F.70600@laposte.net> References: <49CC82DB.1000900@laposte.net> <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> <49CCEBC7.1070807@laposte.net> <191c3a030903270813x1319825bva59b09b789f79855@mail.gmail.com> <49CCFD9F.70600@laposte.net> Message-ID: <191c3a030903271155g751cd9cfk4aab42f60291095c@mail.gmail.com> you have to set disable-transcoding as well to avoid any transcoding situations On Fri, Mar 27, 2009 at 11:23 AM, rod wrote: > Hello, > > I have this error when not enablig proxy_media: > 2009-03-27 19:54:44 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-03-27 19:54:44 [ERR] switch_core_io.c:723 > switch_core_session_write_frame() Codec G.729 decoder error! > > Sure there is an option to check. Any pointers. > > regards. > > > > > Anthony Minessale wrote: > > if you enable mod_g729 you can use freeswitch normally with that g729 > > codec as long > > as no transcoding is enabled (same passthru concept as proxy_media_mode) > > > > > > On Fri, Mar 27, 2009 at 10:07 AM, rod > > wrote: > > > > Hi Brian, > > > > don't understand very well your advice: > > --> Can't use proxy media in this case. (I highly recommend you > > not use > > Proxy Media mode) > > > > If i want to hide my topology network and deal with G729, I must use > > proxy media ? > > Why is Proxy media mode not recommended ?? > > > > regards. > > rod > > > > > > > > Brian West wrote: > > > > > > On Mar 27, 2009, at 2:40 AM, rod wrote: > > > > > >> Hi, > > >> > > >> I did some tests with FS to transcode SIP INFO to RFC2833 (and > vice > > >> versa) and it's working fine when FS stays in the media path with > > >> default configuration. > > >> > > >> But my setup is the following: > > >> - Core network requires SIP INFO > > >> - Peerings require RFC2833 > > >> > > >> all would be fine with FS if my SIP Peers were not enforcing G729 > > >> (discarding G711) so that I have to use the directive > >> application="set" data="proxy_media=true"/> in my dialplan cause > FS > > >> can't deal with G729 except in pass-through. > > > > > > Can't use proxy media in this case. (I highly recommend you not > use > > > Proxy Media mode) > > > > > >> > > >> It's sad, but G729 is still a reality in Telco World. > > > > > > Coming soon! > > > > > >> > > >> So do you think there could be a way to deal with DTMF even if not > > >> analyzing RTP for transcoding. My commercial SBC is doing this, > > but it > > >> sucks and that's the last step before final migration to FS. > > >> > > >> regards, > > >> rod > > > > > > Brian West > > > brian at freeswitch.org > > > > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/5d684059/attachment-0001.html From gkuri at ieee.org Fri Mar 27 11:56:07 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 27 Mar 2009 11:56:07 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> Message-ID: <49CD2147.2070305@ieee.org> I'm not asking what they do, I'm asking those more familiar with FS whether the optimization flags are too aggressive for FS. What do you guys (developers) normalize use, just your basic -march=i686 -pipe ? Gabe Brian West wrote: > Usually if you don't know what they do... then you shouldn't use them! ;) > > > /b > > On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote: > >> regarding gcc compiler optimizations, are they generally compatible with >> FS or should they be removed or does the configure strip them out? just >> curious, as I run Gentoo and use such optimizations as "-march=nocona >> -O2 -pipe -fomit-frame-pointer" >> >> not sure if they break things or I should be removing them before >> compiling FS? >> >> Gabe > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris at fowler.cc Fri Mar 27 12:01:34 2009 From: chris at fowler.cc (Chris Fowler) Date: Fri, 27 Mar 2009 12:01:34 -0700 Subject: [Freeswitch-users] DTMF Missing Digits Message-ID: <1238180494.11750.1307673017@webmail.messagingengine.com> >> Sent: Wednesday, March 25, 2009 12:43 btw you'll have to reinstall your phrase macros .... make vm-sync I think should do it if it doesn't let me know... we removed the 250ms sleeps and that was the problem which we fixed. << I re-did the macros; the only change I could detect was the elimination of the 250ms sleeps; and the change to: I'm running build 12782; should this have fixed it? If so, I will follow the bug reporting instructions you sent earlier. Thanks, Chris. Here's the errors caught today on my production system. 2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '1101' 2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '55' 2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '015' 2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '0000' 2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' From brian at freeswitch.org Fri Mar 27 12:03:29 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 14:03:29 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <49CD2147.2070305@ieee.org> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> <49CD2147.2070305@ieee.org> Message-ID: <0F0D9025-63A1-499C-A9F7-E215BCA20CD8@freeswitch.org> We usually don't specify anything extra! /b On Mar 27, 2009, at 1:56 PM, Gabriel Kuri wrote: > I'm not asking what they do, I'm asking those more familiar with FS > whether the optimization flags are too aggressive for FS. What do you > guys (developers) normalize use, just your basic -march=i686 -pipe ? > > Gabe Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/374cb9c9/attachment.html From anthony.minessale at gmail.com Fri Mar 27 12:04:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 14:04:17 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> Message-ID: <191c3a030903271204s73b96a92k77780034bc7b42be@mail.gmail.com> 1) There is an incompatibility on the fake ncurses wrapper that causes an instant seg fault unless you install the real ncurses. 2) The bleeding edge GCC builds an openzap binary that crashes instantly with no explanation in the core file from a minimal -O2 (that's just the one copmiler bug that we know about for sure, like cock roaches, see one, there are probably 1000) 3) They upgraded to libtool 2.0 which builds binaries that will not start. (easier said than done to upgrade ours too as we have to make sure we work on *every* plarform and the upgrade to make it work would break other operating systems we support) Bottom line, it's not their fault or anything but the choice to use all brand new versions of everything under the sun is not a good idea for your server, it's great that we have bleeding edge stuff or we would not have anyone to test stuff, we have a similar group of people always running SVN trunk of the day. But it's hard to stabalize code when both your code and the OS may be unstable at the same time. There is a reason they call it bleeding vs stable, which one would you rather be if you were in the hospital. =D 2009/3/27 dujinfang > > On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: > > We do not support ubuntu interpid, it has at least 3 known fatal issues not > experienced by all but nonetheless enough to make us unwilling to support > it. > > > I use Ubuntu gutsy in production and interipid in test env. It works well. > Can you briefly explain the 3 fatal issues Anthony? It will help me know > potential risks. > > > > It's "use at your own risk" or use the stable branch "hardy" for any > support. > > > > On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds wrote: > >> Has there been any progress getting FreeSWITCH to build on Ubuntu >> Intrepid without downgrading libtool? >> >> > I successfully built FS on intrepid. I simply done this by changing the > apt-source to Hardy and installed libtool. Obviously I changed the > apt-source back to intrepid after I installed libtool. > > And, another approach. Install libtool from source should be as easy as > configure && make && make install. I done this on a new CentOS4 because the > default yum install of libtool on CentOS4 is old than FS required. > > > Thanks! >> >> Sincerely, >> Trevor Hammonds >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/e22607b1/attachment.html From brian at freeswitch.org Fri Mar 27 12:04:26 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 14:04:26 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238180494.11750.1307673017@webmail.messagingengine.com> References: <1238180494.11750.1307673017@webmail.messagingengine.com> Message-ID: Right and that is the fix for this. If you have the sleep's in your phrase macro's remove them and use the pause= param... you shouldn't have any problems. /b On Mar 27, 2009, at 2:01 PM, Chris Fowler wrote: > > I re-did the macros; the only change I could detect was the > elimination > of the 250ms sleeps; and the change to: > > > I'm running build 12782; should this have fixed it? If so, I will > follow the bug reporting instructions you sent earlier. > > Thanks, Chris. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/f2a095a4/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 27 12:08:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 14:08:23 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <49CD2147.2070305@ieee.org> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> <49CD2147.2070305@ieee.org> Message-ID: <191c3a030903271208v36ffbc95md5311af35dce6e62@mail.gmail.com> We've made no attempts to add any optimization flags on unix to date. We use the defaults and always build debug binaries. When we get some spare time we might go back and turn them on but so far we don't have much of a need to. On Fri, Mar 27, 2009 at 1:56 PM, Gabriel Kuri wrote: > I'm not asking what they do, I'm asking those more familiar with FS > whether the optimization flags are too aggressive for FS. What do you > guys (developers) normalize use, just your basic -march=i686 -pipe ? > > Gabe > > > Brian West wrote: > > Usually if you don't know what they do... then you shouldn't use them! > ;) > > > > > > /b > > > > On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote: > > > >> regarding gcc compiler optimizations, are they generally compatible with > >> FS or should they be removed or does the configure strip them out? just > >> curious, as I run Gentoo and use such optimizations as "-march=nocona > >> -O2 -pipe -fomit-frame-pointer" > >> > >> not sure if they break things or I should be removing them before > >> compiling FS? > >> > >> Gabe > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/f2ebf3bd/attachment.html From gkuri at ieee.org Fri Mar 27 12:11:44 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 27 Mar 2009 12:11:44 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <0F0D9025-63A1-499C-A9F7-E215BCA20CD8@freeswitch.org> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> <49CD2147.2070305@ieee.org> <0F0D9025-63A1-499C-A9F7-E215BCA20CD8@freeswitch.org> Message-ID: <49CD24F0.5090702@ieee.org> ok, thanks, that pretty much amounts to the gcc defaults. perhaps I should recompile FS with those defaults and see if the current jira I have open goes away ;) ... Gabe Brian West wrote: > We usually don't specify anything extra! > > /b > > On Mar 27, 2009, at 1:56 PM, Gabriel Kuri wrote: > >> I'm not asking what they do, I'm asking those more familiar with FS >> whether the optimization flags are too aggressive for FS. What do you >> guys (developers) normalize use, just your basic -march=i686 -pipe ? >> >> Gabe > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Fri Mar 27 12:12:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 14:12:06 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238180494.11750.1307673017@webmail.messagingengine.com> References: <1238180494.11750.1307673017@webmail.messagingengine.com> Message-ID: <191c3a030903271212q270ee6cbv353781981cefb0f2@mail.gmail.com> You should file the bug with the guy who dreamed up RFC2833 ;) Did you provide the menu you are using and what you expect to happen? There are cases where the way you set it up could cause your problems. You also have to realize that dtmf over sip is one of the top 10 gripes ppl have with the protocol and you may have to carefully analyze your traffic since we have zero other complaints open about your problem other than the one from several months ago that brian told you about. 2009/3/27 Chris Fowler > >> > Sent: Wednesday, March 25, 2009 12:43 > > btw you'll have to reinstall your phrase macros .... make vm-sync I > think should do it if it doesn't let me know... we removed the 250ms > sleeps and that was the problem which we fixed. > << > > I re-did the macros; the only change I could detect was the elimination > of the 250ms sleeps; and the change to: > > > I'm running build 12782; should this have fixed it? If so, I will > follow the bug reporting instructions you sent earlier. > > Thanks, Chris. > > > > Here's the errors caught today on my production system. > > 2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '1101' > 2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '55' > 2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '015' > 2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '0000' > 2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > 2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > 2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > 2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > 2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/612f907e/attachment.html From gkuri at ieee.org Fri Mar 27 12:16:35 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 27 Mar 2009 12:16:35 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <191c3a030903271208v36ffbc95md5311af35dce6e62@mail.gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> <49CD2147.2070305@ieee.org> <191c3a030903271208v36ffbc95md5311af35dce6e62@mail.gmail.com> Message-ID: <49CD2613.4000602@ieee.org> Thanks, Anthony. I guess that answers my question, the build process doesn't use whatever is configured locally on the system. Gabe Anthony Minessale wrote: > We've made no attempts to add any optimization flags on unix to date. > We use the defaults and always build debug binaries. > > When we get some spare time we might go back and turn them on but so far > we don't have much of a need to. > > > On Fri, Mar 27, 2009 at 1:56 PM, Gabriel Kuri > wrote: > > I'm not asking what they do, I'm asking those more familiar with FS > whether the optimization flags are too aggressive for FS. What do you > guys (developers) normalize use, just your basic -march=i686 -pipe ? > > Gabe > > > Brian West wrote: > > Usually if you don't know what they do... then you shouldn't use > them! ;) > > > > > > /b > > > > On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote: > > > >> regarding gcc compiler optimizations, are they generally > compatible with > >> FS or should they be removed or does the configure strip them > out? just > >> curious, as I run Gentoo and use such optimizations as "-march=nocona > >> -O2 -pipe -fomit-frame-pointer" > >> > >> not sure if they break things or I should be removing them before > >> compiling FS? > >> > >> Gabe > > > > Brian West > > brian at freeswitch.org > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Prometheus001 at gmx.net Fri Mar 27 12:23:22 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Mar 2009 20:23:22 +0100 Subject: [Freeswitch-users] condition matching on variables which have been set in the dialplan In-Reply-To: <7C44CF00-5B22-4E19-A0A8-74C820E02A46@freeswitch.org> References: <49CCFDE2.4050704@gmx.net> <7C44CF00-5B22-4E19-A0A8-74C820E02A46@freeswitch.org> Message-ID: <49CD27AA.4000503@gmx.net> OK, understood. I will do it in a different way then. Brian West schrieb: > Remember the dialplan is NOT executed when its parsed so you can't set > a var then condition on that exact var on the next line.. that var > doesn't exist. > > /b > > > On Mar 27, 2009, at 11:25 AM, Peter P GMX wrote: > >> >> > continue="true">. >> > data="destination_number=0049xxxxxxxxx"/>. >> > data="destination_number=0049xxxxxxxxx"/>. >> >> >> > Its ${destination_number} > >> expression="^(00[1-9]\d{4,13})$"> >> > data="effective_caller_id_number=unknown"/>. >> > data="effective_caller_id_name=unknown"/>. >> > data="sofia/gateway/QSC_DE/$1 at sip.qsc.de >> "/>. >> >> >> . >> . >> > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Mar 27 12:27:59 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 14:27:59 -0500 Subject: [Freeswitch-users] condition matching on variables which have been set in the dialplan In-Reply-To: <49CD27AA.4000503@gmx.net> References: <49CCFDE2.4050704@gmx.net> <7C44CF00-5B22-4E19-A0A8-74C820E02A46@freeswitch.org> <49CD27AA.4000503@gmx.net> Message-ID: <9019BDB0-C72F-465E-A616-B0B6CDAE9EA9@freeswitch.org> You can execute_extension to revisit the dialplan at a later time kinda like a macro. /b On Mar 27, 2009, at 2:23 PM, Peter P GMX wrote: > OK, understood. I will do it in a different way then. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/5acab493/attachment.html From chris at fowler.cc Fri Mar 27 12:40:08 2009 From: chris at fowler.cc (Chris Fowler) Date: Fri, 27 Mar 2009 12:40:08 -0700 Subject: [Freeswitch-users] DTMF Missing Digits Message-ID: <1238182808.17773.1307679957@webmail.messagingengine.com> >> Did you provide the menu you are using and what you expect to happen? Here's the setup; Caller -> FlowRoute - > FreeSwitch >> B: Right and that is the fix for this. If you have the sleep's in your phrase macro's remove them and use the pause= param... you shouldn't have any problems. Still seeing multiple issues logged during ivr process for mis-interpreted DTMF. Here's today's list from our production server. 2009-03-27 06:38:59 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '1100' 2009-03-27 07:20:33 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '1101' 2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '55' 2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '015' 2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '0000' 2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 11:58:35 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '028' 2009-03-27 11:59:27 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '050' 2009-03-27 12:01:52 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 12:02:01 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 12:02:41 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 12:02:53 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' Any other debug I can capture to assist? Thanks, Chris. From kristian.kielhofner at gmail.com Fri Mar 27 12:47:02 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 27 Mar 2009 15:47:02 -0400 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <09BC8800-D604-40C3-B682-43DFF20DF3B0@freeswitch.org> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <09BC8800-D604-40C3-B682-43DFF20DF3B0@freeswitch.org> Message-ID: <2d9149cd0903271247p37ebf34dib906126b6a9d8d5c@mail.gmail.com> Some platforms add/use URI params in the user portion of the URI. I was just reminded of this this week with Global Crossing: sip:+18005551212;npdi=yes at 0.0.0.0 That is (of course) the number portability display indicator... ;) Anyways, I had to rewrite the URI using OpenSIPS to strip the ";npdi=yes". I imagine you could do the same but I'm not sure how in FreeSWITCH... 2009/3/27 Brian West : > Params are not added before the @ sign that I'm pretty sure of. > data="sip_invite_params=this=kewl;that=iskewl"/> > > > > Results in?INVITE sip:blah at blah;this=kewl;that=iskewl SIP/2.0 > > You also have sip_invite_to_params, sip_invite_from_params, > sip_invite_contact_params > /b > > > On Mar 27, 2009, at 5:11 AM, James H Thompson wrote: > > I need to generate calls with Invite URIs in this format: > > INVITE?sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060?SIP/2.0 > > Is there an easy way to do this? > > Thanks. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Fri Mar 27 12:48:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 27 Mar 2009 15:48:50 -0400 Subject: [Freeswitch-users] Contacting Callie Message-ID: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> What is the best way (if any) to contact "Callie" for custom prompt work? I can't seem to find much about her. Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Fri Mar 27 12:49:16 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 14:49:16 -0500 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <2d9149cd0903271247p37ebf34dib906126b6a9d8d5c@mail.gmail.com> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <09BC8800-D604-40C3-B682-43DFF20DF3B0@freeswitch.org> <2d9149cd0903271247p37ebf34dib906126b6a9d8d5c@mail.gmail.com> Message-ID: Thats about dumb... um I don't think we let you put things in there.. but it would only be a few lines to do it...blah rfc's suck! /b On Mar 27, 2009, at 2:47 PM, Kristian Kielhofner wrote: > Some platforms add/use URI params in the user portion of the URI. > > I was just reminded of this this week with Global Crossing: > > sip:+18005551212;npdi=yes at 0.0.0.0 > > That is (of course) the number portability display indicator... ;) > > Anyways, I had to rewrite the URI using OpenSIPS to strip the > ";npdi=yes". I imagine you could do the same but I'm not sure how in > FreeSWITCH. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/3829c479/attachment.html From jht at lj.net Fri Mar 27 12:52:30 2009 From: jht at lj.net (James H Thompson) Date: Fri, 27 Mar 2009 09:52:30 -1000 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> Message-ID: <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> Calls would be sent to the IP address after the '@' in the URI. Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as the user part of a SIP URI. My example Invite URI is the way we are receiving traffic from some of the major telecom carriers. We would like be able to generate calls using the same formats. ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Friday, March 27, 2009 5:41 AM Subject: Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls You seem to be confusing your standards, those 2 specs are about tel: uri's not sip: uris. Sending a tel uri I am not sure we can do, where would we send it to? Mike On Mar 27, 2009, at 6:11 AM, James H Thompson wrote: I need to generate calls with Invite URIs in this format: INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 SIP/2.0 Is there an easy way to do this? Thanks. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/6b5775de/attachment-0001.html From anthony.minessale at gmail.com Fri Mar 27 13:05:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 15:05:07 -0500 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> Message-ID: <191c3a030903271305m68c07a2fh767c3f5b2499aa@mail.gmail.com> if you prefix the sofia dial string with sip: you should be able to pass anything you want. sofia/internal/sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 2009/3/27 James H Thompson > Calls would be sent to the IP address after the '@' <%27@%27> in the > URI. > Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as the > user part of a SIP URI. > My example Invite URI is the way we are receiving traffic from some of the > major telecom carriers. > We would like be able to generate calls using the same formats. > > > ----- Original Message ----- *From:* Michael Jerris > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Friday, March 27, 2009 5:41 AM > *Subject:* Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls > > You seem to be confusing your standards, those 2 specs are about tel: uri's > not sip: uris. Sending a tel uri I am not sure we can do, where would we > send it to? > Mike > > On Mar 27, 2009, at 6:11 AM, James H Thompson wrote: > > I need to generate calls with Invite URIs in this format: > > INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 > SIP/2.0 > > Is there an easy way to do this? > > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/acbdb383/attachment.html From william.suffill at gmail.com Fri Mar 27 13:04:51 2009 From: william.suffill at gmail.com (William Suffill) Date: Fri, 27 Mar 2009 16:04:51 -0400 Subject: [Freeswitch-users] Contacting Callie In-Reply-To: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> References: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> Message-ID: <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> Good question. Last I talked to Brian about this (new prompts for upcoming new release) all the prompts are done by http://www.gmvoices.com/. I don't know anything more about the process to get recordings done or if there is any preferred process if they are from users of FreeSwitch but be curious to find out. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/8ac119f0/attachment.html From anthony.minessale at gmail.com Fri Mar 27 13:06:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 15:06:44 -0500 Subject: [Freeswitch-users] Contacting Callie In-Reply-To: <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> References: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> Message-ID: <191c3a030903271306x37a851d1mcd90295f1837a574@mail.gmail.com> you can get the through us, just contact brian 2009/3/27 William Suffill > Good question. Last I talked to Brian about this (new prompts for upcoming > new release) all the prompts are done by http://www.gmvoices.com/. I don't > know anything more about the process to get recordings done or if there is > any preferred process if they are from users of FreeSwitch but be curious to > find out. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/be48521d/attachment.html From brian at freeswitch.org Fri Mar 27 13:11:19 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 15:11:19 -0500 Subject: [Freeswitch-users] Contacting Callie In-Reply-To: <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> References: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> Message-ID: <1B17EFD7-AB3D-44B5-BDE4-9C6B82B8BFA1@freeswitch.org> Well I do get a discount if we batch them. I'm taking donations for this order brian at freeswitch.org is my paypal we are sending the order out monday but I only have a handful of stuff to record this go around. http://jira.freeswitch.org/browse/FSSCRIPTS-15 William, Thanks for your donation to help pay for it. ;) /b On Mar 27, 2009, at 3:04 PM, William Suffill wrote: > Good question. Last I talked to Brian about this (new prompts for > upcoming new release) all the prompts are done by http://www.gmvoices.com/ > . I don't know anything more about the process to get recordings > done or if there is any preferred process if they are from users of > FreeSwitch but be curious to find out. > > -- W Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/e792f639/attachment.html From freeswitch at servercorps.com Fri Mar 27 13:14:42 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Fri, 27 Mar 2009 15:14:42 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> Message-ID: <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> Also, moving the list to Google Groups would allow email OR threaded views, and personally I like them better than nabble. anm_ On Thu, Mar 26, 2009 at 11:06 AM, Michael Jerris wrote: > http://n2.nabble.com/freeswitch-users-f2379917.html > > Mike > > > On Mar 26, 2009, at 12:01 PM, Tim Ringenbach wrote: > >> Is there nothing out there that integrates a forum with a mailing >> list? >> It seems like one could display the mailing list archives exactly >> like a >> forum, and allow users to register to the forum and post (appearing to >> the mailing list as username at forumurl.org) in such a way that they >> don't >> have to realize it's a mailing list. >> >> Anthony Minessale wrote: >>> The guy started a forum almost a month ago and as you can see nobody >>> knows the url and it has no posts. >>> >>> http://freeswitch411.info/forum/ >>> >>> This is one of the problems I was worried about when endorsing a >>> forum. >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rupa at rupa.com Fri Mar 27 13:22:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 27 Mar 2009 15:22:23 -0500 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CCEC15.8010500@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> <49CB9E0C.4030300@ewetel.de> <49CCEC15.8010500@ewetel.de> Message-ID: kill -9 on a thread will kill the process which kills freeswitch. On Fri, Mar 27, 2009 at 10:09 AM, Helmut Kuper wrote: > Hello, > > today I killed that special thread via "kill -9" a simple kill didn't > helped. Unfortunately this led to a normal shutdown of FS although I > killed not the parent process. :( > > After restart of FS the server has a normal load again. > > regards and a nice weekend > Helmut > > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/8ab9bb89/attachment-0001.html From jht at lj.net Fri Mar 27 13:43:42 2009 From: jht at lj.net (James H Thompson) Date: Fri, 27 Mar 2009 10:43:42 -1000 Subject: [Freeswitch-users] IRC is not for all References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com><3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> Message-ID: <091e01c9af1c$b7a44130$0aa8a8c0@jhtp28> The freeswitch user mailing list is also on: http://news.gmane.org/gmane.comp.telephony.freeswitch.user There are several forums packages that allow feeding in a mailing list, although not many people seem to do it. Google Groups and Yahoo Groups are also alternatives. I've been considering mirroring some of the major voip mailing lists on voip-info.org into a forum of somekind. If this would be of interest let me know. Jim ----- Original Message ----- From: "Addison Martin" To: Sent: Friday, March 27, 2009 10:14 AM Subject: Re: [Freeswitch-users] IRC is not for all > Also, moving the list to Google Groups would allow email OR threaded > views, and personally I like them better than nabble. > > anm_ > > > On Thu, Mar 26, 2009 at 11:06 AM, Michael Jerris wrote: >> http://n2.nabble.com/freeswitch-users-f2379917.html >> >> Mike >> >> >> On Mar 26, 2009, at 12:01 PM, Tim Ringenbach wrote: >> >>> Is there nothing out there that integrates a forum with a mailing >>> list? >>> It seems like one could display the mailing list archives exactly >>> like a >>> forum, and allow users to register to the forum and post (appearing to >>> the mailing list as username at forumurl.org) in such a way that they >>> don't >>> have to realize it's a mailing list. >>> >>> Anthony Minessale wrote: >>>> The guy started a forum almost a month ago and as you can see nobody >>>> knows the url and it has no posts. >>>> >>>> http://freeswitch411.info/forum/ >>>> >>>> This is one of the problems I was worried about when endorsing a >>>> forum. >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Fri Mar 27 18:42:55 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 28 Mar 2009 12:42:55 +1100 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <091e01c9af1c$b7a44130$0aa8a8c0@jhtp28> References: <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> <091e01c9af1c$b7a44130$0aa8a8c0@jhtp28> Message-ID: <20090328014255.GA10819@jdc.jasonjgw.net> James H Thompson wrote: > I've been considering mirroring some of the major voip mailing lists on > voip-info.org > into a forum of somekind. Have a look at http://www.gmane.org/ and note that you can post via NNTP or via the WEb. This mailing list is subscribed to gmane. From jason at jasonjgw.net Fri Mar 27 18:47:18 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 28 Mar 2009 12:47:18 +1100 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> Message-ID: <20090328014718.GA10860@jdc.jasonjgw.net> Addison Martin wrote: > Also, moving the list to Google Groups would allow email OR threaded > views, and personally I like them better than nabble. http://dir.gmane.org/gmane.comp.telephony.freeswitch.user Would any of those views suffice? From dujinfang at gmail.com Fri Mar 27 19:22:23 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 28 Mar 2009 10:22:23 +0800 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <191c3a030903271204s73b96a92k77780034bc7b42be@mail.gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <191c3a030903271204s73b96a92k77780034bc7b42be@mail.gmail.com> Message-ID: <0D693D25-7BC3-4D22-8D05-6103BADEB31A@gmail.com> Thanks. I was thinking created a new server with Ubuntu intrepid, seems I'd like back to Hardy. Even on hardy the default libtool is version 2. replace to libtool 1 should be easy as I mentioned before. On Mar 28, 2009, at 3:04 AM, Anthony Minessale wrote: > 1) There is an incompatibility on the fake ncurses wrapper that > causes an instant seg fault unless you install the real ncurses. On ubuntu it's libncurses5-dev, use for simple's sake. > > 2) The bleeding edge GCC builds an openzap binary that crashes > instantly with no explanation in the core file from a minimal -O2 > (that's just the one copmiler bug that we know about for sure, like > cock roaches, see one, there are probably 1000) we don't use openzap. Is the probably 1000 all in the openzap or anywhere else potentially?. > > 3) They upgraded to libtool 2.0 which builds binaries that will not > start. (easier said than done to upgrade ours too as we have to make > sure we work on *every* plarform and the upgrade to make it work > would break other operating systems we support) > Understand. > Bottom line, it's not their fault or anything but the choice to use > all brand new versions of everything under the sun is not a good > idea for your server, it's great that we have bleeding edge stuff or > we would not have anyone to test stuff, we have a similar group of > people always running SVN trunk of the day. But it's hard to > stabalize code when both your code and the OS may be unstable at the > same time. > > There is a reason they call it bleeding vs stable, which one would > you rather be if you were in the hospital. =D > As you mentioned. It's not their fault. ppl want to live on the edge just need to install multi-versions of gcc(or other tools). Like the Linux kernel, to compile from source, gcc-3 was recommended for a long time. Don't know if it's still the case recently. > > > 2009/3/27 dujinfang > > On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: >> We do not support ubuntu interpid, it has at least 3 known fatal >> issues not experienced by all but nonetheless enough to make us >> unwilling to support it. > > I use Ubuntu gutsy in production and interipid in test env. It > works well. Can you briefly explain the 3 fatal issues Anthony? It > will help me know potential risks. > >> >> >> It's "use at your own risk" or use the stable branch "hardy" for >> any support. >> >> >> >> On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds > > wrote: >> Has there been any progress getting FreeSWITCH to build on Ubuntu >> Intrepid without downgrading libtool? >> > > I successfully built FS on intrepid. I simply done this by changing > the apt-source to Hardy and installed libtool. Obviously I changed > the apt-source back to intrepid after I installed libtool. > > And, another approach. Install libtool from source should be as easy > as configure && make && make install. I done this on a new CentOS4 > because the default yum install of libtool on CentOS4 is old than FS > required. > > >> Thanks! >> >> Sincerely, >> Trevor Hammonds >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/9ab6bfca/attachment.html From dujinfang at gmail.com Fri Mar 27 19:39:25 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 28 Mar 2009 10:39:25 +0800 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <191c3a030903271305m68c07a2fh767c3f5b2499aa@mail.gmail.com> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> <191c3a030903271305m68c07a2fh767c3f5b2499aa@mail.gmail.com> Message-ID: <7F119E06-EC37-41EF-9919-45D2B9358880@gmail.com> On Mar 28, 2009, at 4:05 AM, Anthony Minessale wrote: > if you prefix the sofia dial string with sip: you should be able to > pass anything you want. > > sofia/internal/sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 > Is that similar as this? got it from wiki: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#From_the_Dialplan > 2009/3/27 James H Thompson > Calls would be sent to the IP address after the '@' in the URI. > Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as > the user part of a SIP URI. > My example Invite URI is the way we are receiving traffic from some > of the major telecom carriers. > We would like be able to generate calls using the same formats. > > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Friday, March 27, 2009 5:41 AM > Subject: Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls > > You seem to be confusing your standards, those 2 specs are about > tel: uri's not sip: uris. Sending a tel uri I am not sure we can > do, where would we send it to? > > Mike > > On Mar 27, 2009, at 6:11 AM, James H Thompson wrote: > >> I need to generate calls with Invite URIs in this format: >> >> INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 >> SIP/2.0 >> >> Is there an easy way to do this? >> >> Thanks. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/4de5879c/attachment-0001.html From brian at freeswitch.org Fri Mar 27 19:55:41 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 21:55:41 -0500 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <7F119E06-EC37-41EF-9919-45D2B9358880@gmail.com> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> <191c3a030903271305m68c07a2fh767c3f5b2499aa@mail.gmail.com> <7F119E06-EC37-41EF-9919-45D2B9358880@gmail.com> Message-ID: no sofia/profile/sip:blah at blah sip: makes sofia take it as is. /b On Mar 27, 2009, at 9:39 PM, dujinfang wrote: > > Is that similar as this? > > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/eeb3b0fb/attachment.html From hads at nice.net.nz Fri Mar 27 21:23:47 2009 From: hads at nice.net.nz (Hadley Rich) Date: Sat, 28 Mar 2009 17:23:47 +1300 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <0D693D25-7BC3-4D22-8D05-6103BADEB31A@gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903271204s73b96a92k77780034bc7b42be@mail.gmail.com> <0D693D25-7BC3-4D22-8D05-6103BADEB31A@gmail.com> Message-ID: <200903281723.47626.hads@nice.net.nz> On Sat, 28 Mar 2009 15:22:23 dujinfang wrote: > Even on hardy the default libtool is version 2. replace to libtool 1 > should be easy as I mentioned before. The libtool on Hardy is 1.5.26 hads -- http://nicegear.co.nz VoIP, DVB and other Linux compatible hardware. From moizchinoy at gmail.com Sat Mar 28 02:09:02 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Sat, 28 Mar 2009 13:09:02 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <1B117EFC18604C6E82663FDAEB3342A8@quos> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> Message-ID: <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> I am trying it on windows. Where to get gnutls-devel for windows? On Fri, Mar 27, 2009 at 6:20 AM, Andrea wrote: > try to install ?gnutls-devel (name on centos) before to compile. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. From dave at 3c.co.uk Sat Mar 28 06:44:49 2009 From: dave at 3c.co.uk (David Knell) Date: Sat, 28 Mar 2009 07:44:49 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C0F9EA.3000200@freeswitch.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> Message-ID: <49CE29D1.3000603@3c.co.uk> Raymond Chandler wrote: > What's interesting to me is.... everyone on this thread except you has > said that in real-world scenarios, they need the EC for reliability. > One of which, does signal processing programming professionally. It > seems to me that if you "build a better mouse trap" you must know what's > involved in making it work properly. I'm not sure what your background > really is, but you'd be hard pressed to match up to Steve's reputation > and/or experience. > Public willy-waving is undignified but, in brief, I've built and sold IVRs since 1997, wrote a CAPI-based soft IVR in 1999 (which required software for, inter alia, DTMF detection), developed a software fax modem (V.29, V.27ter, T.30, etc.) which I sold to a CTI card vendor and so on. I've collected some data, of which it is commonly said that the plural of anecdote - which is what we've had so far - is not. The IVR collects a 16 digit DTMF string, terminated by #. TDM->IP conversion was performed by an Asterisk box with an el-cheapo quad E1 card (no EC) for half the calls, and an AS5400 (with EC) for the other half. The proportion of entries missing one or more digit was 3.1% (Asterisk) and 3.3% (AS5400); this is not a statistically significant difference given the sample size. The reason for looking at this criterion is (a) that it's easy to measure, and (b) the most likely way that a DTMF detector will fail in the presence of excess noise, which includes echo, would be to miss a digit. This error rate is the sum of human error + detector error, and I've no measurements to show how this might be split; I would expect it's almost all human. Note that this is a digit error rate of about 1 in 500. This is, of course, only data from one site, but it's a start; it's only by collecting data such as this that one can understand how well one's mouse trap works and whether it needs improvement or not. > That said, it might be a good idea to just agree to disagree as this is > starting to sound like the faxing over IP talks I hear a lot. (i.e. > "faxing over g.711u with no t.38 works fine for me") Where it might work > for some people by some mysterious phenomena, it won't work for the > general public. And telling people that they don't need EC, where so > many have already said that they obviously do, is just as irresponsible, > IMHO, as you claiming Steve was for telling them that they don't need it. > That's a simplification. Simple IVR (record, replay, collect DTMF) probably doesn't need EC; if you're trying to do ASR with barge-in, bridge callers to other callers or operators, etc., then you probably do. I am interested that the recommended solution is 'buy Sangoma' - expensive and proprietary - when Oslec, a FOSS echo cancellers which, by all accounts, works extremely well, is out there and has been for some time. --Dave From mike at jerris.com Sat Mar 28 09:03:08 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 28 Mar 2009 12:03:08 -0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> Message-ID: <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> please see my previous response in this thread. MIke On Mar 28, 2009, at 5:09 AM, Moiz Chinoy wrote: > I am trying it on windows. > Where to get gnutls-devel for windows? > From mszlazak at aol.com Sat Mar 28 10:58:20 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 28 Mar 2009 13:58:20 -0400 Subject: [Freeswitch-users] switch_ivr_originate() Parse Error! Message-ID: <8CB7DE5EA49B71E-A84-2D59@WEBMAIL-MY06.sysops.aol.com> I'm getting a parsing error which seems to be coming from the space in "Extension 1000". If this is "normal" then what's the best way to deal with spaces in caller id names? This is how the call was originated: ??? ??? ??? 2009-03-28 10:50:56 [ERR] switch_ivr_originate.c:976 switch_ivr_originate() Parse Error! 2009-03-28 10:50:56 [DEBUG] switch_ivr_originate.c:2081 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2009-03-28 10:50:56 [DEBUG] mod_commands.c:2213 sch_api_callback() Command originate({id_name=Extension 1000,id_number=1000}sofia/internal/1000%10.0.0.3 GINO_ANS): -ERR DESTINATION_OUT_OF_ORDER Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/ff950b85/attachment.html From brian at freeswitch.org Sat Mar 28 11:13:44 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Mar 2009 13:13:44 -0500 Subject: [Freeswitch-users] switch_ivr_originate() Parse Error! In-Reply-To: <8CB7DE5EA49B71E-A84-2D59@WEBMAIL-MY06.sysops.aol.com> References: <8CB7DE5EA49B71E-A84-2D59@WEBMAIL-MY06.sysops.aol.com> Message-ID: <303F2923-BD29-4924-BE54-BBAFF47C139F@freeswitch.org> quote it! Single quote '${blah}' /b On Mar 28, 2009, at 12:58 PM, mszlazak at aol.com wrote: > I'm getting a parsing error which seems to be coming from the space > in "Extension 1000". If this is "normal" then what's the best way to > deal with spaces in caller id names? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/3a272c5b/attachment.html From bipin at xbipin.com Sat Mar 28 12:21:37 2009 From: bipin at xbipin.com (xbipin) Date: Sat, 28 Mar 2009 12:21:37 -0700 (PDT) Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> Message-ID: <22760426.post@talk.nabble.com> hi, im trying to do the same, use FS as a plain and simple SBC but cant figure out how to do so -- View this message in context: http://www.nabble.com/freeswitch-as-a-session-border-controller-tp16755838p22760426.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Sat Mar 28 15:35:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 28 Mar 2009 17:35:35 -0500 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49CE29D1.3000603@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> Message-ID: <191c3a030903281535o6f3d09dbye20cf947cd250d80@mail.gmail.com> Well, >From my experience, an AS5300 thinks nearly *anthing* is dtmf. The one I have as a PSTN onramp to our conference bridge drives me bonkers with false positives since every key on the pad means something in our default conference. Jim Dixon designed the el-cheapo (tormenta 2) to put as many resources on the host as possible, that was the goal behind the initiative. The driver for this card was the template on which the entire Zaptel (now Dahdi) was based. What's interesting is that early zapata library (a user space abstraction lib) was completely consumed by chan_zap and from there may of the featues were gravitated towards the linux kernel then eventually into the hardware as new cards were developed. The typical reason for this kind of evolution is customers. When there was no TDM to be had at all, el-cheapo and software was the bees knees. As they started getting more greedy and anxious for higher quality, they started asking for improvements that lead to more stuff onboard in the new cards. I am guessing this will continue until the cards are too expensive and we will go full circle back to all-on-host just in time for the 16 core CPU box being standard issue. It all depends on the strategy employed, if you want to use a tor2 and oslec, (a software echo canceller that, in fact has none other than Steve Underwood from this thread as a collaberator) then do it. If it ain't broke, don't fix it.....If you prefer to have hardware EC, then buy a card that supports it. ------ see http://www.rowetel.com/ucasterisk/oslec.html Background and Credits Oslec started life as a prototype echo canceller and G168 test framework from Steve Underwood's Spandsp library. Steve wrote much of the DSP code used in Asterisk, and the Zaptel echo cancellation code is heavily based on his work. ------ Bottom line: There is no real correct answer because it depends on what your goals are and what you personally prefer. I personally have used both, I am annoyed with hardware EC because it breaks software dtmf but now the sangoma drivers have hardware dtmf to use together with hardware EC so that solves the problem. I would prefer not to take any sides in this debate since everyone on this thread has contributed greatly to our project and I respect them all. I would, however, like to ask that maybe we can channel all of this intelligence into some common goal and do somethng great rather than spend our energy doing the techno-geek version of m&m freestyle rapping. On Sat, Mar 28, 2009 at 8:44 AM, David Knell wrote: > Raymond Chandler wrote: > > What's interesting to me is.... everyone on this thread except you has > > said that in real-world scenarios, they need the EC for reliability. > > One of which, does signal processing programming professionally. It > > seems to me that if you "build a better mouse trap" you must know what's > > involved in making it work properly. I'm not sure what your background > > really is, but you'd be hard pressed to match up to Steve's reputation > > and/or experience. > > > Public willy-waving is undignified but, in brief, I've built and sold > IVRs since > 1997, wrote a CAPI-based soft IVR in 1999 (which required software for, > inter > alia, DTMF detection), developed a software fax modem (V.29, V.27ter, T.30, > etc.) which I sold to a CTI card vendor and so on. > > I've collected some data, of which it is commonly said that the plural > of anecdote - > which is what we've had so far - is not. The IVR collects a 16 digit > DTMF string, > terminated by #. TDM->IP conversion was performed by an Asterisk box with > an el-cheapo quad E1 card (no EC) for half the calls, and an AS5400 > (with EC) > for the other half. > > The proportion of entries missing one or more digit was 3.1% (Asterisk) > and 3.3% > (AS5400); this is not a statistically significant difference given the > sample size. > The reason for looking at this criterion is (a) that it's easy to > measure, and (b) the > most likely way that a DTMF detector will fail in the presence of excess > noise, > which includes echo, would be to miss a digit. This error rate is the > sum of > human error + detector error, and I've no measurements to show how this > might > be split; I would expect it's almost all human. Note that this is a > digit error rate of > about 1 in 500. > > This is, of course, only data from one site, but it's a start; it's only > by collecting > data such as this that one can understand how well one's mouse trap works > and whether it needs improvement or not. > > That said, it might be a good idea to just agree to disagree as this is > > starting to sound like the faxing over IP talks I hear a lot. (i.e. > > "faxing over g.711u with no t.38 works fine for me") Where it might work > > for some people by some mysterious phenomena, it won't work for the > > general public. And telling people that they don't need EC, where so > > many have already said that they obviously do, is just as irresponsible, > > IMHO, as you claiming Steve was for telling them that they don't need it. > > > That's a simplification. Simple IVR (record, replay, collect DTMF) > probably > doesn't need EC; if you're trying to do ASR with barge-in, bridge callers > to > other callers or operators, etc., then you probably do. > > I am interested that the recommended solution is 'buy Sangoma' - expensive > and proprietary - when Oslec, a FOSS echo cancellers which, by all > accounts, > works extremely well, is out there and has been for some time. > > --Dave > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/c79e865f/attachment-0001.html From saeedahmad1981 at gmail.com Sat Mar 28 16:49:53 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Sun, 29 Mar 2009 00:49:53 +0100 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <22760426.post@talk.nabble.com> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> Message-ID: <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> http://wiki.freeswitch.org/wiki/SBC_Setup written by rod. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of xbipin Sent: Saturday, March 28, 2009 8:22 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a session border controller hi, im trying to do the same, use FS as a plain and simple SBC but cant figure out how to do so -- View this message in context: http://www.nabble.com/freeswitch-as-a-session-border-controller-tp16755838p2 2760426.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From f.koliqi at gmail.com Sat Mar 28 22:30:36 2009 From: f.koliqi at gmail.com (Fadil Berisha) Date: Sun, 29 Mar 2009 01:30:36 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49CE29D1.3000603@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> Message-ID: <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> > That's a simplification. Simple IVR (record, replay, collect DTMF) > probably > doesn't need EC; > Dave Collect DTMF does not need EC. I take out your word "probably" because no need for any dilemma. Interaction between DTMF detector an EC *when EC exist * is different question and deserve separate thread. Although I am confirming your statement,** I can not say "I am voting for you", simple because this is not political forum to express believing to one or other leader or authority. With respect koliqi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090329/689b231b/attachment.html From steveu at coppice.org Sun Mar 29 01:32:56 2009 From: steveu at coppice.org (Steve Underwood) Date: Sun, 29 Mar 2009 16:32:56 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> Message-ID: <49CF3238.9040503@coppice.org> Fadil Berisha wrote: > > That's a simplification. Simple IVR (record, replay, collect > DTMF) probably > doesn't need EC; > > > > Dave > > Collect DTMF does not need EC. I take out your word "probably" > because no need for any dilemma. Interaction between DTMF detector an > EC *when EC exist* is different question and deserve separate thread. > Although I am confirming your statement,// I can not say "I am voting > for you", simple because this is not political forum to express > believing to one or other leader or authority. Receiving DTMF reliably needs a signal to noise ratio of about 10dB if the noise is Gaussian. The statistics of voice mean you need the DTMF to be more like 15dB above voice. Most hybrids are only required to have a return loss of better than 12dB. They can be *much* better, but don't count on it, especially at the phone's hybrid. You have multiple hybrids in the path (usually 2 or 4). Let's take the better case with only 2 sitting between the outgoing exchange card and the far end phone. Put 10km of copper between the exchange the phone (typical copper planning limit) and you probably have 15dB of attenuation on the line. Now your DTMF is actually below the level of the voice prompt for much of the time. Think that will work with an echo canceller? Sure you can get reliable DTMF detection on 70%-80% of call paths with no echo cancellation, but if you want reliability with close to 100% of phone lines, you need echo cancellation to remove the voice prompt from the signal received at the IVR. Dialogic, NMS, and the others didn't put EC on their cards for nothing. The only reason the normal connection of a phone to a line card gets reliable detection of the first dialed digit in the presence of a dialing tone is that the DTMF detector heavily filters that dialing tone. Forget the political forum crap. If you want to refute what I just said, try to back up your argument with some actual engineering. Regards, Steve From bipin at xbipin.com Sat Mar 28 22:50:52 2009 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 29 Mar 2009 09:50:52 +0400 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> Message-ID: <49CF0C3C.8070508@xbipin.com> hi, what are the other steps if done on windows platform with FS as SBC and Voipswitch as the server. secondly how to make the clients registering to voipswitch go through FS in which case FS should simply accept all registrations but it should do a forward registration to voipswitch and if voipswitch rejects then FS should also reject. basically i need FS as a SBC with topology hiding. Regards, Bipin www.xbipin.com +971-55-9270058 -------- Original Message -------- Subject: Re: [Freeswitch-users] freeswitch as a session border controller From: Saeed Ahmed To: freeswitch-users at lists.freeswitch.org Date: Sunday, March 29, 2009 3:49:53 AM > http://wiki.freeswitch.org/wiki/SBC_Setup written by rod. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of xbipin > Sent: Saturday, March 28, 2009 8:22 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] freeswitch as a session border controller > > > hi, > > im trying to do the same, use FS as a plain and simple SBC but cant figure > out how to do so From bipin at xbipin.com Sat Mar 28 23:03:34 2009 From: bipin at xbipin.com (xbipin) Date: Sat, 28 Mar 2009 23:03:34 -0700 (PDT) Subject: [Freeswitch-users] how to do upper registration Message-ID: <22743580.post@talk.nabble.com> i have been trying to do this but seem to have been lost, actually i want to use freeswitch as a session border controller so basically all the clients that try to register to FS will actually be authenticated voipswitch but i want FS to be in between and proxy everything and with any of its advanced options turned off. -- View this message in context: http://www.nabble.com/how-to-do-upper-registration-tp22743580p22743580.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From bipin at xbipin.com Sat Mar 28 23:07:02 2009 From: bipin at xbipin.com (xbipin) Date: Sat, 28 Mar 2009 23:07:02 -0700 (PDT) Subject: [Freeswitch-users] upper registration in FS? Message-ID: <22764757.post@talk.nabble.com> i have been trying to do this but seem to have been lost, actually i want to use freeswitch as a session border controller so basically all the clients that try to register to FS will actually be authenticated by voipswitch but i want FS to be in between and proxy everything and with any of its advanced options turned off so as to simply work as topology hiding, proxy, SBC only. -- View this message in context: http://www.nabble.com/upper-registration-in-FS--tp22764757p22764757.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From krice at suspicious.org Sun Mar 29 12:07:23 2009 From: krice at suspicious.org (Ken Rice) Date: Sun, 29 Mar 2009 14:07:23 -0500 Subject: [Freeswitch-users] how to do upper registration In-Reply-To: <22743580.post@talk.nabble.com> Message-ID: FreeSwitch is NOT a proxy... It is a B2BUA... You need to look at something like OpenSIPS/OpenSER for what you are trying to do. > From: xbipin > Reply-To: > Date: Sat, 28 Mar 2009 23:03:34 -0700 (PDT) > To: > Subject: [Freeswitch-users] how to do upper registration > > > i have been trying to do this but seem to have been lost, actually i want to > use freeswitch as a session border controller so basically all the clients > that try to register to FS will actually be authenticated voipswitch but i > want FS to be in between and proxy everything and with any of its advanced > options turned off. > -- > View this message in context: > http://www.nabble.com/how-to-do-upper-registration-tp22743580p22743580.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From grevenx at me.com Sun Mar 29 14:16:13 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Sun, 29 Mar 2009 23:16:13 +0200 Subject: [Freeswitch-users] how to do upper registration In-Reply-To: References: Message-ID: While it's not a proxy, FS CAN be a SBC as he mentions (though I'm not sure the feature he describes fits the bill for a SBC?). http://en.wikipedia.org/wiki/Back-to-back_user_agent http://en.wikipedia.org/wiki/Session_Border_Controller http://wiki.freeswitch.org/wiki/Specsheet Best regards, Even Andr? On 29. mars. 2009, at 21.07, Ken Rice wrote: > FreeSwitch is NOT a proxy... It is a B2BUA... You need to look at > something > like OpenSIPS/OpenSER for what you are trying to do. > > > > >> From: xbipin >> Reply-To: >> Date: Sat, 28 Mar 2009 23:03:34 -0700 (PDT) >> To: >> Subject: [Freeswitch-users] how to do upper registration >> >> >> i have been trying to do this but seem to have been lost, actually >> i want to >> use freeswitch as a session border controller so basically all the >> clients >> that try to register to FS will actually be authenticated >> voipswitch but i >> want FS to be in between and proxy everything and with any of its >> advanced >> options turned off. >> -- >> View this message in context: >> http://www.nabble.com/how-to-do-upper-registration-tp22743580p22743580.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave.cabot at iconsoluciones.net Sun Mar 29 13:16:26 2009 From: dave.cabot at iconsoluciones.net (Dave Cabot) Date: Sun, 29 Mar 2009 16:16:26 -0400 Subject: [Freeswitch-users] mod_skypiax Message-ID: <1238357786.7199.29.camel@dave-laptop> Hi folks. I've been playing with mod_skypiax. 1) it seems that the documentation web pages are out of date. http://jira.freeswitch.org/ rejects connections. I'm running FreeSWITCH Version 1.0.trunk (12826). I'm using Xvfb and skype 2.0.0.72-1_i386.deb. I can make a call and sound is great, but then when the call ends, so does skype. Attached is what gdb puked: root at freeswitch-1:/usr/local/freeswitch/conf# gdb /usr/bin/skype 20250 GNU gdb 6.8-debian Copyright (C) 2008 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "i486-linux-gnu"... (no debugging symbols found) Attaching to program: /usr/bin/skype, process 20250 Reading symbols from /usr/lib/libasound.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libasound.so.2 Reading symbols from /usr/lib/libXv.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXv.so.1 Reading symbols from /usr/lib/libXss.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXss.so.1 Reading symbols from /lib/tls/i686/cmov/librt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/librt.so.1 Reading symbols from /usr/lib/libQtDBus.so.4... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtDBus.so.4 Reading symbols from /usr/lib/libQtGui.so.4...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtGui.so.4 Reading symbols from /usr/lib/libQtNetwork.so.4... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtNetwork.so.4 Reading symbols from /usr/lib/libQtCore.so.4...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtCore.so.4 Reading symbols from /lib/tls/i686/cmov/libpthread.so.0... (no debugging symbols found)...done. [Thread debugging using libthread_db enabled] [New Thread 0xb6ef16c0 (LWP 20250)] [New Thread 0xb5708b90 (LWP 20261)] [New Thread 0xb6116b90 (LWP 20259)] [New Thread 0xb6197b90 (LWP 20256)] [New Thread 0xb6218b90 (LWP 20255)] [New Thread 0xb6a19b90 (LWP 20254)] [New Thread 0xb6a9ab90 (LWP 20253)] [New Thread 0xb6db3b90 (LWP 20251)] Loaded symbols for /lib/tls/i686/cmov/libpthread.so.0 Reading symbols from /usr/lib/libstdc++.so.6...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libstdc++.so.6 Reading symbols from /lib/tls/i686/cmov/libm.so.6... (no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libm.so.6 Reading symbols from /lib/libgcc_s.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libgcc_s.so.1 Reading symbols from /lib/tls/i686/cmov/libc.so.6... (no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libc.so.6 Reading symbols from /usr/lib/libX11.so.6...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libX11.so.6 Reading symbols from /usr/lib/libXext.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXext.so.6 Reading symbols from /lib/tls/i686/cmov/libdl.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libdl.so.2 Reading symbols from /lib/ld-linux.so.2... (no debugging symbols found)...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/lib/libdbus-1.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libdbus-1.so.3 Reading symbols from /usr/lib/libQtXml.so.4... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtXml.so.4 Reading symbols from /usr/lib/libfontconfig.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libfontconfig.so.1 Reading symbols from /usr/lib/libz.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libz.so.1 Reading symbols from /usr/lib/libgthread-2.0.so.0...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libgthread-2.0.so.0 Reading symbols from /usr/lib/libglib-2.0.so.0... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libglib-2.0.so.0 Reading symbols from /usr/lib/libaudio.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libaudio.so.2 Reading symbols from /usr/lib/libXt.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXt.so.6 Reading symbols from /usr/lib/libpng12.so.0...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libpng12.so.0 Reading symbols from /usr/lib/libSM.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libSM.so.6 Reading symbols from /usr/lib/libICE.so.6...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libICE.so.6 Reading symbols from /usr/lib/libXi.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXi.so.6 Reading symbols from /usr/lib/libXrender.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXrender.so.1 Reading symbols from /usr/lib/libXrandr.so.2... ---Type to continue, or q to quit--- (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXrandr.so.2 Reading symbols from /usr/lib/libXfixes.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXfixes.so.3 Reading symbols from /usr/lib/libXcursor.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXcursor.so.1 Reading symbols from /usr/lib/libXinerama.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXinerama.so.1 Reading symbols from /usr/lib/libfreetype.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libfreetype.so.6 Reading symbols from /usr/lib/libxcb-xlib.so.0...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libxcb-xlib.so.0 Reading symbols from /usr/lib/libxcb.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libxcb.so.1 Reading symbols from /usr/lib/libXau.so.6...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXau.so.6 Reading symbols from /usr/lib/libexpat.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libexpat.so.1 Reading symbols from /usr/lib/libpcre.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libpcre.so.3 Reading symbols from /usr/lib/libXdmcp.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXdmcp.so.6 Reading symbols from /usr/lib/gconv/UTF-16.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/gconv/UTF-16.so Reading symbols from /usr/lib/qt4/plugins/imageformats/libqgif.so... (no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqgif.so Reading symbols from /usr/lib/qt4/plugins/imageformats/libqjpeg.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqjpeg.so Reading symbols from /usr/lib/libjpeg.so.62... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libjpeg.so.62 Reading symbols from /usr/lib/qt4/plugins/imageformats/libqmng.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqmng.so Reading symbols from /usr/lib/libmng.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libmng.so.1 Reading symbols from /usr/lib/liblcms.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/liblcms.so.1 Reading symbols from /usr/lib/qt4/plugins/imageformats/libqsvg.so... (no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqsvg.so Reading symbols from /usr/lib/libQtSvg.so.4...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtSvg.so.4 Reading symbols from /usr/lib/qt4/plugins/imageformats/libqtiff.so... (no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqtiff.so Reading symbols from /usr/lib/i686/cmov/libssl.so.0.9.8...(no debugging symbols found)...done. Loaded symbols for /usr/lib/i686/cmov/libssl.so.0.9.8 Reading symbols from /usr/lib/i686/cmov/libcrypto.so.0.9.8... (no debugging symbols found)...done. Loaded symbols for /usr/lib/i686/cmov/libcrypto.so.0.9.8 Reading symbols from /usr/lib/libresolv.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libresolv.so Reading symbols from /lib/tls/i686/cmov/libnss_files.so.2... (no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libnss_files.so.2 Reading symbols from /lib/tls/i686/cmov/libnss_dns.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libnss_dns.so.2 (no debugging symbols found) 0xb7fdc410 in __kernel_vsyscall () (gdb) c Continuing. [New Thread 0xb6093b90 (LWP 20278)] [New Thread 0xb4eeab90 (LWP 20279)] [New Thread 0xb46e9b90 (LWP 20280)] RtApiAlsa: callback thread error (RtApiAlsa: audio write error for device (Default device (default)): Exec format error.) ... closing stream. [Thread 0xb4eeab90 (LWP 20279) exited] Skype Xv: Xv ports available: 0 Skype XShm: XShm support enabled [New Thread 0xb3e9cb90 (LWP 20283)] RtApiAlsa: callback thread error (RtApiAlsa: audio read error for device (Default device (default)): Unknown error 405.) ... closing stream. [Thread 0xb46e9b90 (LWP 20280) exited] [Thread 0xb3e9cb90 (LWP 20283) exited] [New Thread 0xb3d3db90 (LWP 20288)] Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 0xb6197b90 (LWP 20256)] 0xb738c4ac in free () from /lib/tls/i686/cmov/libc.so.6 (gdb) Continuing. Program received signal SIGABRT, Aborted. 0xb7fdc410 in __kernel_vsyscall () (gdb) bt #0 0xb7fdc410 in __kernel_vsyscall () #1 0xb7348085 in raise () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7349a01 in abort () from /lib/tls/i686/cmov/libc.so.6 #3 0x082f372a in ?? () #4 #5 0xb738c4ac in free () from /lib/tls/i686/cmov/libc.so.6 #6 0x08808972 in ?? () Backtrace stopped: previous frame inner to this frame (corrupt stack?) (gdb) quit The program is running. Quit anyway (and detach it)? (y or n) y Detaching from program: /usr/bin/skype, process 20250 root at freeswitch-1:/usr/local/freeswitch/conf# -bash: line 320: 20249 Done echo "blah blah" 20250 Aborted (core dumped) | DISPLAY=:101 /usr/bin/skype --pipelogin [1]+ Exit 134 ( echo "blah blah" | DISPLAY=:101 /usr/bin/skype --pipelogin ) Not only do I have to restart skype, but I also have to reload mod_skypiax. Obviously this has it's drawbacks. Has anyone seen this and have a solution tucked away? Thanks From can_man at gmx.de Sun Mar 29 15:09:21 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Mon, 30 Mar 2009 00:09:21 +0200 Subject: [Freeswitch-users] FS - MjSip no voice Message-ID: <20090329220921.227090@gmx.net> Hello everyone, I am trying to get FS working with the MjSip Java Sip-stack, the SipToSis source and the normal one. Everything works well within my own network and when using x-lite, but when it comes to making calls from MjSip to an outside FS server I don't hear any voice - seems to be a NAT problem or some kind of other MjSip problem. Registration works fine though and SIP messages get through ok, but non of the UDP RTP ones. Would be great if someone could advice me on how to do the setup correctly. The whole FS trace can be found here: http://pastebin.freeswitch.org/8029 The settings for MjSip are: "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090", "transport_protocols=udp tcp","from_url=", "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", "bin_rat=rat","bin_vic=vic" Thank you very much. Best wishes, Phil -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From brian at freeswitch.org Sun Mar 29 16:50:21 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 29 Mar 2009 18:50:21 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax In-Reply-To: <1238357786.7199.29.camel@dave-laptop> References: <1238357786.7199.29.camel@dave-laptop> Message-ID: <5C65F009-BAC1-40EF-BD69-46547F267346@freeswitch.org> Please redirect that to jira. I have fixed it. /b On Mar 29, 2009, at 3:16 PM, Dave Cabot wrote: > 1) it seems that the documentation web pages are out of date. > http://jira.freeswitch.org/ rejects connections. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090329/6d709f04/attachment.html From f.koliqi at gmail.com Sun Mar 29 19:20:20 2009 From: f.koliqi at gmail.com (Fadil Berisha) Date: Sun, 29 Mar 2009 22:20:20 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49CF3238.9040503@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> <49CF3238.9040503@coppice.org> Message-ID: <5c7d82f20903291920od56993eu7a30a760e0ed4847@mail.gmail.com> > Sure you can get reliable DTMF detection on 70%-80% of call paths with > no echo cancellation, Fair enough. You say "sure you can get reliable DTMF detection on 70%-80% of call paths with no echo cancellation". OK, you forgot to mention that this is achievable with text-book algorithms and that exist advanced algorithms with reliability close to 100%. From my point of view , no need further arguing this issue and this thread for me is closed. With respect koliqi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090329/dac37785/attachment.html From christian.bourke1 at gmail.com Sun Mar 29 17:55:14 2009 From: christian.bourke1 at gmail.com (Christian Bourke) Date: Mon, 30 Mar 2009 10:55:14 +1000 Subject: [Freeswitch-users] H323 supported devices Message-ID: <564d7da90903291755x1caf83e7k4c2e74a2761ba24c@mail.gmail.com> Hi, I would like to implement freeswitch as a H323 gateway. I would like to be able to receive calls from H323 devices such as Polycom V500 and Dlink DVC-1000. Does anyone know if this is possible? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/2f144d24/attachment.html From krice at freeswitch.org Sun Mar 29 22:58:58 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 30 Mar 2009 00:58:58 -0500 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: <564d7da90903291755x1caf83e7k4c2e74a2761ba24c@mail.gmail.com> Message-ID: Yes its possible... Look at mod_opal From: Christian Bourke Reply-To: Date: Mon, 30 Mar 2009 10:55:14 +1000 To: Subject: [Freeswitch-users] H323 supported devices Hi, ? I would like to implement freeswitch as a H323 gateway. I would like to be able to receive calls from H323 devices such as Polycom V500 and Dlink DVC-1000. ? Does anyone know if this is possible? ? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/97589ec2/attachment.html From moizchinoy at gmail.com Mon Mar 30 00:56:46 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Mon, 30 Mar 2009 11:56:46 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> Message-ID: <29b888f80903300056t51828330s39bbf662fe2cd8c0@mail.gmail.com> Hi, If there is no build for the tls with freeswitch on windows, can you please guide how to test gtalk integration on windows. I have successfully compiled the tls with iksemel, there were couple of errors but I managed to compile it. Now, I am also getting the error libdingaling.c:1545 xmpp_connect() io error 2 7 What is this error about? There is a similar post on the mailing list regarding the above IO error but can't find the reply for it. On Sat, Mar 28, 2009 at 8:03 PM, Michael Jerris wrote: > please see my previous response in this thread. > > MIke > > On Mar 28, 2009, at 5:09 AM, Moiz Chinoy wrote: > >> I am trying it on windows. >> Where to get gnutls-devel for windows? >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. From christian.bourke1 at gmail.com Mon Mar 30 01:28:08 2009 From: christian.bourke1 at gmail.com (Christian Bourke) Date: Mon, 30 Mar 2009 18:28:08 +1000 Subject: [Freeswitch-users] Freeswitch Hunt Group Message-ID: <564d7da90903300128q70196f17r4834ec7ffb933fcd@mail.gmail.com> Hi, I would like to implement freeswitch as an H323 gatekeeper. I would like to know if its possible for incoming H323 calls to go into a hunt group, if no 'operators' are logged into the hunt group then the call will bounce to an auto attendant and the H323 caller will be played a short video message then freeswitch will disconnect the call. Does this sound possible with Freeswitch? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/6802b65d/attachment.html From christian.bourke1 at gmail.com Mon Mar 30 01:30:49 2009 From: christian.bourke1 at gmail.com (Christian Bourke) Date: Mon, 30 Mar 2009 18:30:49 +1000 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: References: <564d7da90903291755x1caf83e7k4c2e74a2761ba24c@mail.gmail.com> Message-ID: <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> Thank you Ken. Do you know if mod_opal supports both H323 voice and video? 2009/3/30 Ken Rice > Yes its possible... Look at mod_opal > > > > ------------------------------ > *From: *Christian Bourke > *Reply-To: * > *Date: *Mon, 30 Mar 2009 10:55:14 +1000 > *To: * > *Subject: *[Freeswitch-users] H323 supported devices > > Hi, > > I would like to implement freeswitch as a H323 gateway. I would like to be > able to receive calls from H323 devices such as Polycom V500 and Dlink > DVC-1000. > > Does anyone know if this is possible? > > > ------------------------------ > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/8aee4f2c/attachment.html From fdelawarde at wirelessmundi.com Mon Mar 30 01:34:10 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 30 Mar 2009 10:34:10 +0200 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues In-Reply-To: <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> References: <1238155504.4364.222.camel@localhost.localdomain> <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> Message-ID: <1238402050.4364.274.camel@localhost.localdomain> Thanks for your quick&clear answers. On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote: > If you are using on-hook agents, it will place as many outbound calls > as there are people waiting. > If you are using off-hook agents it will just connect the first free > agent. By "people waiting" you mean "calls in the queue" and not "agents waiting" right? > > - Is there an [easy] way (with some javascript or similar) to > "emulate" Asterisk's distribution strategies to agents (by > amount of time without calls, total number of answered calls, > round robing, ...) in this cases? > > > Easiest way would be to write a patch in C to mod_fifo it'self or > propose a bounty for features and see if you can get the change > approved by the developers. I'll give the patch a try once I have a bit more practice with FreeSWITCH (never even launched it yet) and its API. Any hints or "quick-start" guides for module writing? Fran?ois. From monemran at gmail.com Mon Mar 30 01:34:34 2009 From: monemran at gmail.com (M.Emran) Date: Mon, 30 Mar 2009 14:34:34 +0600 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> References: <564d7da90903291755x1caf83e7k4c2e74a2761ba24c@mail.gmail.com> <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> Message-ID: How is the performance for mod_opal ? is the performance same as SIP ? 2009/3/30 Christian Bourke > Thank you Ken. Do you know if mod_opal supports both H323 voice and video? > > 2009/3/30 Ken Rice > >> Yes its possible... Look at mod_opal >> >> >> >> ------------------------------ >> *From: *Christian Bourke >> *Reply-To: * >> *Date: *Mon, 30 Mar 2009 10:55:14 +1000 >> *To: * >> *Subject: *[Freeswitch-users] H323 supported devices >> >> Hi, >> >> I would like to implement freeswitch as a H323 gateway. I would like to be >> able to receive calls from H323 devices such as Polycom V500 and Dlink >> DVC-1000. >> >> Does anyone know if this is possible? >> >> >> ------------------------------ >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards ---------- M Emran Managing Director InSpiration Software Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/fadb1f3c/attachment.html From krice at suspicious.org Mon Mar 30 03:40:13 2009 From: krice at suspicious.org (Ken Rice) Date: Mon, 30 Mar 2009 05:40:13 -0500 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> Message-ID: I?m not sure if the video support is complete but both freeswitch and opal support video... Now I also see another email you send asking about freeswitch being a gatekeeper... FreeSWITCH is just another h323 end point, it does not do all gatekeeper functions like keep up with what IP what h323 user is on... However, as on your other message it might be possible to do what you want... My suggestion would be to set it up and try... There needs to be more h323 testing anyway Ken From: Christian Bourke Reply-To: Date: Mon, 30 Mar 2009 18:30:49 +1000 To: Subject: Re: [Freeswitch-users] H323 supported devices Thank you Ken. Do you know if?mod_opal supports both H323 voice and video? 2009/3/30 Ken Rice > Yes its possible... Look at mod_opal > > > > > From: Christian Bourke > Reply-To: > Date: Mon, 30 Mar 2009 10:55:14 +1000 > To: > Subject: [Freeswitch-users] H323 supported devices > > > Hi, > ? > I would like to implement freeswitch as a H323 gateway. I would like to be > able to receive calls from H323 devices such as Polycom V500 and Dlink > DVC-1000. > ? > Does anyone know if this is possible? > ? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/441e43bc/attachment.html From bipin at xbipin.com Sun Mar 29 22:28:11 2009 From: bipin at xbipin.com (xbipin) Date: Sun, 29 Mar 2009 22:28:11 -0700 (PDT) Subject: [Freeswitch-users] how to do upper registration In-Reply-To: References: <22743580.post@talk.nabble.com> Message-ID: <22776631.post@talk.nabble.com> so what i make of it is that FS cant do upper registration or lets say it wont proxy everything and wont use the registration details it receives from clients registering to it and use it to register clients to a different server for which its proxying, or mayb its not meant to do so but then i ask can FS be configured to accept all registrations coming to it without each userid and password being defined in the profiles or can FS be configured to simply accept all incomming calls without the need for clients to register to it in which case it will fullfill my needs partially. or one more thing is use the id and pass sent by client to be forwarded to my actual server and then its up to the server to authenticate it and connect the call or to reply with incorrect id and pass. this can be done only if FS could accept all registrations without it being defined in its profile and some how the dialplan was crafted to use the registration id and pass sent by client to be used as the id and pass for the SIP TRUNK or the server in general. -- View this message in context: http://www.nabble.com/how-to-do-upper-registration-tp22770442p22776631.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From bipin at xbipin.com Mon Mar 30 01:21:39 2009 From: bipin at xbipin.com (xbipin) Date: Mon, 30 Mar 2009 01:21:39 -0700 (PDT) Subject: [Freeswitch-users] how to do upper registration Message-ID: <22776631.post@talk.nabble.com> so what i make of it is that FS cant do upper registration or lets say it wont proxy everything and wont use the registration details it receives from clients registering to it and use it to register clients to a different server for which its proxying, or mayb its not meant to do so but then i ask can FS be configured to accept all registrations coming to it without each userid and password being defined in the profiles or can FS be configured to simply accept all incomming calls without the need for clients to register to it in which case it will fullfill my needs partially. or one more thing is use the id and pass sent by client to be forwarded to my actual server and then its up to the server to authenticate it and connect the call or to reply with incorrect id and pass. this can be done only if FS could accept all registrations without it being defined in its profile and some how the dialplan was crafted to use the registration id and pass sent by client to be used as the id and pass for the SIP TRUNK or the server in general. -- View this message in context: http://www.nabble.com/how-to-do-upper-registration-tp22770442p22776631.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From steveu at coppice.org Mon Mar 30 04:48:03 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 30 Mar 2009 19:48:03 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <5c7d82f20903291920od56993eu7a30a760e0ed4847@mail.gmail.com> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> <49CF3238.9040503@coppice.org> <5c7d82f20903291920od56993eu7a30a760e0ed4847@mail.gmail.com> Message-ID: <49D0B173.9080002@coppice.org> Fadil Berisha wrote: > > Sure you can get reliable DTMF detection on 70%-80% of call paths with > no echo cancellation, > > > Fair enough. You say "sure you can get reliable DTMF detection on > 70%-80% of call paths with no echo cancellation". OK, you forgot to > mention that this is achievable with text-book algorithms and that > exist advanced algorithms with reliability close to 100%. From my > point of view , no need further arguing this issue and this thread for > me is closed. Could you enlighten us as to these advanced algorithms that beat conventional statistics? You keep referring to mystical adaptive algorithms on the OSLEC mailing list, but you never give any details there, either. Steve From anthony.minessale at gmail.com Mon Mar 30 05:53:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 07:53:39 -0500 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <22764757.post@talk.nabble.com> References: <22764757.post@talk.nabble.com> Message-ID: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> FS does not proxy registrations, really it doesn't proxy anything very much besides calls in a limited capacity, you would want to use a SIP proxy for that. On Sun, Mar 29, 2009 at 1:07 AM, xbipin wrote: > > i have been trying to do this but seem to have been lost, actually i want > to > use freeswitch as a session border controller so basically all the clients > that try to register to FS will actually be authenticated by voipswitch but > i want FS to be in between and proxy everything and with any of its > advanced > options turned off so as to simply work as topology hiding, proxy, SBC > only. > -- > View this message in context: > http://www.nabble.com/upper-registration-in-FS--tp22764757p22764757.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/29a99765/attachment-0001.html From anthony.minessale at gmail.com Mon Mar 30 06:04:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:04:31 -0500 Subject: [Freeswitch-users] FS - MjSip no voice In-Reply-To: <20090329220921.227090@gmx.net> References: <20090329220921.227090@gmx.net> Message-ID: <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> You should press f8 to get more detailed output from FS. you also should capture more of the call, starting at line 192 you seem to be sending yourself a notify, not sure how you did that. you are not by any chance trying to call a registered endpoint using the FS ip together with @ are you? say you fs box is 1.2.3.4 and the phone is registered as 1000 If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you would use sofia/internal/1000%1.2.3.4 The % tells it to resolve the domain as a locally hosted domain and translate it to the registered contact instead of using dns. otherwise, enable debugging with f8 and reproduce your issue and capture *all* the output. On Sun, Mar 29, 2009 at 5:09 PM, wrote: > Hello everyone, > > I am trying to get FS working with the MjSip Java Sip-stack, the SipToSis > source and the normal one. Everything works well within my own network and > when using x-lite, but when it comes to making calls from MjSip to an > outside FS server I don't hear any voice - seems to be a NAT problem or some > kind of other MjSip problem. Registration works fine though and SIP messages > get through ok, but non of the UDP RTP ones. Would be great if someone could > advice me on how to do the setup correctly. > > The whole FS trace can be found here: http://pastebin.freeswitch.org/8029 > > The settings for MjSip are: > > "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090", > "transport_protocols=udp tcp","from_url=", > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > "bin_rat=rat","bin_vic=vic" > > > Thank you very much. > Best wishes, > Phil > > -- > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/017be9a0/attachment.html From anthony.minessale at gmail.com Mon Mar 30 06:07:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:07:19 -0500 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <29b888f80903300056t51828330s39bbf662fe2cd8c0@mail.gmail.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> <29b888f80903300056t51828330s39bbf662fe2cd8c0@mail.gmail.com> Message-ID: <191c3a030903300607m55d7dcd8u78e8ea9fc7b491cb@mail.gmail.com> libdingaling.c:1545 xmpp_connect() io error 2 7 It means there was an I/O error trying to connect it's probably because iksemel does not have srv support, if you are trying to connect to gtalk you have to manually specify the server as talk.google.com, the default client.xml is self-explanatory for connecting to gtalk and it's on the wiki too.. On Mon, Mar 30, 2009 at 2:56 AM, Moiz Chinoy wrote: > Hi, > > If there is no build for the tls with freeswitch on windows, can you > please guide how to test gtalk integration on windows. > > I have successfully compiled the tls with iksemel, there were couple > of errors but I managed to compile it. > > Now, I am also getting the error libdingaling.c:1545 xmpp_connect() io > error 2 7 > > What is this error about? > > There is a similar post on the mailing list regarding the above IO > error but can't find the reply for it. > > On Sat, Mar 28, 2009 at 8:03 PM, Michael Jerris wrote: > > please see my previous response in this thread. > > > > MIke > > > > On Mar 28, 2009, at 5:09 AM, Moiz Chinoy wrote: > > > >> I am trying it on windows. > >> Where to get gnutls-devel for windows? > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Regards, > Moiz Chinoy. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/01eeb0ac/attachment.html From anthony.minessale at gmail.com Mon Mar 30 06:12:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:12:05 -0500 Subject: [Freeswitch-users] how to do upper registration In-Reply-To: <22776631.post@talk.nabble.com> References: <22776631.post@talk.nabble.com> Message-ID: <191c3a030903300612g19fb50f6n3b10ba7e8f650282@mail.gmail.com> is this deja-vu, this is the 2nd thread with the same subject, i seem to be seeing multiple of every email you send you may want to check your mail client. like ken said, and me too in the other thread with the same exact email in it, use a proxy http://www.kamailio.org/ On Mon, Mar 30, 2009 at 3:21 AM, xbipin wrote: > > so what i make of it is that FS cant do upper registration or lets say it > wont proxy everything and wont use the registration details it receives > from > clients registering to it and use it to register clients to a different > server for which its proxying, or mayb its not meant to do so but then i > ask > can FS be configured to accept all registrations coming to it without each > userid and password being defined in the profiles or can FS be configured > to > simply accept all incomming calls without the need for clients to register > to it in which case it will fullfill my needs partially. > > or one more thing is use the id and pass sent by client to be forwarded to > my actual server and then its up to the server to authenticate it and > connect the call or to reply with incorrect id and pass. this can be done > only if FS could accept all registrations without it being defined in its > profile and some how the dialplan was crafted to use the registration id > and > pass sent by client to be used as the id and pass for the SIP TRUNK or the > server in general. > -- > View this message in context: > http://www.nabble.com/how-to-do-upper-registration-tp22770442p22776631.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/4217d439/attachment.html From anthony.minessale at gmail.com Mon Mar 30 06:13:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:13:36 -0500 Subject: [Freeswitch-users] Freeswitch Hunt Group In-Reply-To: <564d7da90903300128q70196f17r4834ec7ffb933fcd@mail.gmail.com> References: <564d7da90903300128q70196f17r4834ec7ffb933fcd@mail.gmail.com> Message-ID: <191c3a030903300613y11efe2bcyb69e46740eba5e45@mail.gmail.com> the h323 support in FS is brand new. It's lacking video support at all. You would have to provide funding to round up some developers to implement what you need. 2009/3/30 Christian Bourke > Hi, > > I would like to implement freeswitch as an H323 gatekeeper. I would like to > know if its possible for incoming H323 calls to go into a hunt group, if no > 'operators' are logged into the hunt group then the call will bounce to an > auto attendant and the H323 caller will be played a short video message then > freeswitch will disconnect the call. > > Does this sound possible with Freeswitch? > > Thanks, > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/91ed57d5/attachment-0001.html From anthony.minessale at gmail.com Mon Mar 30 06:14:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:14:35 -0500 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues In-Reply-To: <1238402050.4364.274.camel@localhost.localdomain> References: <1238155504.4364.222.camel@localhost.localdomain> <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> <1238402050.4364.274.camel@localhost.localdomain> Message-ID: <191c3a030903300614g25257ba4i4b46e5ab0caffe7b@mail.gmail.com> On Mon, Mar 30, 2009 at 3:34 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > Thanks for your quick&clear answers. > > > On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote: > > > If you are using on-hook agents, it will place as many outbound calls > > as there are people waiting. > > If you are using off-hook agents it will just connect the first free > > agent. > > By "people waiting" you mean "calls in the queue" and not "agents > waiting" right? > correct > > > > > - Is there an [easy] way (with some javascript or similar) to > > "emulate" Asterisk's distribution strategies to agents (by > > amount of time without calls, total number of answered calls, > > round robing, ...) in this cases? > > > > > > Easiest way would be to write a patch in C to mod_fifo it'self or > > propose a bounty for features and see if you can get the change > > approved by the developers. > > I'll give the patch a try once I have a bit more practice with > FreeSWITCH (never even launched it yet) and its API. Any hints or > "quick-start" guides for module writing? > > Fran?ois. > > ok, stop by irc if you need any pointers > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/8324ece5/attachment.html From anthony.minessale at gmail.com Mon Mar 30 06:15:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:15:25 -0500 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: References: <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> Message-ID: <191c3a030903300615m74a3e301pbeac813f41ada8c3@mail.gmail.com> perhaps we should stick to one thread per topic within the same 1 week span. 2009/3/30 Ken Rice > I?m not sure if the video support is complete but both freeswitch and > opal support video... > Now I also see another email you send asking about freeswitch being a > gatekeeper... FreeSWITCH is just another h323 end point, it does not do all > gatekeeper functions like keep up with what IP what h323 user is on... > > However, as on your other message it might be possible to do what you > want... My suggestion would be to set it up and try... There needs to be > more h323 testing anyway > > Ken > > > ------------------------------ > *From: *Christian Bourke > *Reply-To: * > *Date: *Mon, 30 Mar 2009 18:30:49 +1000 > *To: * > *Subject: *Re: [Freeswitch-users] H323 supported devices > > Thank you Ken. Do you know if mod_opal supports both H323 voice and video? > > 2009/3/30 Ken Rice > > Yes its possible... Look at mod_opal > > > > ------------------------------ > *From: *Christian Bourke > *Reply-To: * > *Date: *Mon, 30 Mar 2009 10:55:14 +1000 > *To: * > *Subject: *[Freeswitch-users] H323 supported devices > > > Hi, > > I would like to implement freeswitch as a H323 gateway. I would like to be > able to receive calls from H323 devices such as Polycom V500 and Dlink > DVC-1000. > > Does anyone know if this is possible? > > > ------------------------------ > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/898ae964/attachment.html From anthony.minessale at gmail.com Mon Mar 30 06:24:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:24:49 -0500 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <49CF0C3C.8070508@xbipin.com> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> <49CF0C3C.8070508@xbipin.com> Message-ID: <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> I'm really starting to feel like we're playing musical threads here. On Sun, Mar 29, 2009 at 12:50 AM, Bipin Patel wrote: > hi, > > what are the other steps if done on windows platform with FS as SBC and > Voipswitch as the server. > secondly how to make the clients registering to voipswitch go through FS > in which case FS should simply accept all registrations but it should do > a forward registration to voipswitch and if voipswitch rejects then FS > should also reject. basically i need FS as a SBC with topology hiding. > > Regards, > Bipin > www.xbipin.com > +971-55-9270058 > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] freeswitch as a session border controller > From: Saeed Ahmed > To: freeswitch-users at lists.freeswitch.org > Date: Sunday, March 29, 2009 3:49:53 AM > > > http://wiki.freeswitch.org/wiki/SBC_Setup written by rod. > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > xbipin > > Sent: Saturday, March 28, 2009 8:22 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] freeswitch as a session border controller > > > > > > hi, > > > > im trying to do the same, use FS as a plain and simple SBC but cant > figure > > out how to do so > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/97dcccca/attachment.html From steveu at coppice.org Mon Mar 30 06:56:18 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 30 Mar 2009 21:56:18 +0800 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> <49CF0C3C.8070508@xbipin.com> <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> Message-ID: <49D0CF82.7060009@coppice.org> Anthony Minessale wrote: > I'm really starting to feel like we're playing musical threads here. Just avoid playing them through low bit rate codecs. :-) Steve From dave at 3c.co.uk Mon Mar 30 07:14:21 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 30 Mar 2009 08:14:21 -0600 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <49D0CF82.7060009@coppice.org> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> <49CF0C3C.8070508@xbipin.com> <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> <49D0CF82.7060009@coppice.org> Message-ID: <49D0D3BD.5070602@3c.co.uk> Steve Underwood wrote: > Anthony Minessale wrote: > >> I'm really starting to feel like we're playing musical threads here. >> > Just avoid playing them through low bit rate codecs. :-) > I think we need an echo canceller ;-) --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/985933e8/attachment.html From freeswitch at servercorps.com Mon Mar 30 07:17:00 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Mon, 30 Mar 2009 09:17:00 -0500 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <49D0D3BD.5070602@3c.co.uk> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> <49CF0C3C.8070508@xbipin.com> <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> <49D0CF82.7060009@coppice.org> <49D0D3BD.5070602@3c.co.uk> Message-ID: <92e7d2090903300717m2e620117m1653624ed19a8eea@mail.gmail.com> GAAAAHH /me gouges eyes out with EC card 2009/3/30 David Knell : > Steve Underwood wrote: > > Anthony Minessale wrote: > > > I'm really starting to feel like we're playing musical threads here. > > > Just avoid playing them through low bit rate codecs. :-) > > > I think we need an echo canceller ;-) > > --Dave > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From bipin at xbipin.com Mon Mar 30 07:33:26 2009 From: bipin at xbipin.com (xbipin) Date: Mon, 30 Mar 2009 07:33:26 -0700 (PDT) Subject: [Freeswitch-users] live iso image with freeswitch Message-ID: <22784622.post@talk.nabble.com> can any1 tell me where can i find a live cd image with the basic stuff to run FS and FS with all it tools installed and WITH A GUI, something like a pbx in a flash iso image so windows users like me find it easier to get testing with FS as the support for windows SIP proxy or any SIP related tool for windows platform is just about nil so i realized FS on windows also wont make much sense coz the rest of the developers etc use linux for FS and if i simply keep waiting for FS to actually do something productive on windows platform then it might take long or forever. if any i can provide me a live CD image with just enough tools to run FS to its fullest coz till date i have been only using Voipswitch and its time i need to implement TLS or any such type of encryption to reach new markets. -- View this message in context: http://www.nabble.com/live-iso-image-with-freeswitch-tp22784622p22784622.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Mar 30 07:43:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 09:43:27 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22784622.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> Message-ID: <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> On Mar 30, 2009, at 9:33 AM, xbipin wrote: > can any1 tell me where can i find a live cd image with the basic > stuff to run > FS and FS with all it tools installed and WITH A GUI, something like > a pbx > in a flash iso image so windows users like me find it easier to get > testing > with FS as the support for windows SIP proxy or any SIP related tool > for > windows platform is just about nil so i realized FS on windows also > wont > make much sense coz the rest of the developers etc use linux for FS > and if i > simply keep waiting for FS to actually do something productive on > windows > platform then it might take long or forever. Well this is a tall order... You know people ask for it.. or shall I say demand it... but nobody really steps up to help out at all on the GUI requests. FreeSWITCH on windows is equally capable minus a couple of things like TLS since nobody will actually DO the work required to make it happen. This isn't a buffet where you pull up and demand things be one way or the other... this is a community where you start helping. I would love to see more helping and less demanding! > if any i can provide me a live CD image with just enough tools to > run FS to > its fullest coz till date i have been only using Voipswitch and its > time i > need to implement TLS or any such type of encryption to reach new > markets. You could follow the linux how to and install CentOS and be done already. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/11d6eac6/attachment.html From gmaruzz at celliax.org Mon Mar 30 07:44:01 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 16:44:01 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22784622.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> Message-ID: <7b197bef0903300744p7b388be2x2ce1b30841722e3@mail.gmail.com> There is none yet available. If you have patience, I suspect that one will be out in the next weeks, tough. Watch the website and the mailing list for announcement. 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 Mon, Mar 30, 2009 at 4:33 PM, xbipin wrote: > > can any1 tell me where can i find a live cd image with the basic stuff to run > FS and FS with all it tools installed and WITH A GUI, something like a pbx > in a flash iso image so windows users like me find it easier to get testing > with FS as the support for windows SIP proxy or any SIP related tool for > windows platform is just about nil so i realized FS on windows also wont > make much sense coz the rest of the developers etc use linux for FS and if i > simply keep waiting for FS to actually do something productive on windows > platform then it might take long or forever. > > if any i can provide me a live CD image with just enough tools to run FS to > its fullest coz till date i have been only using Voipswitch and its time i > need to implement TLS or any such type of encryption to reach new markets. > -- > View this message in context: http://www.nabble.com/live-iso-image-with-freeswitch-tp22784622p22784622.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Mon Mar 30 07:44:58 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 30 Mar 2009 10:44:58 -0400 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22784622.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> Message-ID: On Mar 30, 2009, at 10:33 AM, xbipin wrote: > > can any1 tell me where can i find a live cd image with the basic > stuff to run > FS and FS with all it tools installed and WITH A GUI, something like > a pbx > in a flash iso image so windows users like me find it easier to get > testing > with FS as the support for windows SIP proxy or any SIP related tool > for > windows platform is just about nil so i realized FS on windows also > wont > make much sense coz the rest of the developers etc use linux for FS > and if i > simply keep waiting for FS to actually do something productive on > windows > platform then it might take long or forever. There is not currently a fully working gui or a live cd image. I think your comments about FreeSWITCH on windows are not correct. > if any i can provide me a live CD image with just enough tools to > run FS to > its fullest coz till date i have been only using Voipswitch and its > time i > need to implement TLS or any such type of encryption to reach new > markets. There is a live cd in the works, I am sure it will be announced here when it is ready. Mike From msc at freeswitch.org Mon Mar 30 08:23:59 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 30 Mar 2009 08:23:59 -0700 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues In-Reply-To: <1238402050.4364.274.camel@localhost.localdomain> References: <1238155504.4364.222.camel@localhost.localdomain> <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> <1238402050.4364.274.camel@localhost.localdomain> Message-ID: On Mar 30, 2009, at 1:34 AM, Fran?ois Delawarde wrote: > Thanks for your quick&clear answers. > > > On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote: > >> If you are using on-hook agents, it will place as many outbound calls >> as there are people waiting. >> If you are using off-hook agents it will just connect the first free >> agent. > > By "people waiting" you mean "calls in the queue" and not "agents > waiting" right? > >> >> - Is there an [easy] way (with some javascript or similar) to >> "emulate" Asterisk's distribution strategies to agents (by >> amount of time without calls, total number of answered calls, >> round robing, ...) in this cases? >> >> >> Easiest way would be to write a patch in C to mod_fifo it'self or >> propose a bounty for features and see if you can get the change >> approved by the developers. > > I'll give the patch a try once I have a bit more practice with > FreeSWITCH (never even launched it yet) and its API. Any hints or > "quick-start" guides for module writing? > There are some developer docs on the wiki. Nothing truly comprehensive but enough to get you going. Also see mod_skel in the source tree. -MC From bipin at xbipin.com Mon Mar 30 09:18:06 2009 From: bipin at xbipin.com (xbipin) Date: Mon, 30 Mar 2009 09:18:06 -0700 (PDT) Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> Message-ID: <22786890.post@talk.nabble.com> hi, im really sorry if it sounded as if i was demanding, its a place where every1 helps so i did the same thing, asked if any1 knew of any such live cd image coz its been quiet some time and i havent reached anywhere with the testing of FS coz the reason being, i can test it offline at my home but that will be like a single client connecting to FS and again the issue being, i live in a country where VoIP is blocked so cant even get FS to register to a provider also. then how do i ever test it, so in reply i would say i have a dedicated server and a proper VoIP setup with loads of clients but i have been using Voipswitch on windows till date and they got a very simple form of encryption which helps me bypass blocks and do the selling as well as testing, so now if i install FS on my server then how can i still test it in my existing setup coz the TLS, as discussed, is currently not functional on the windows platform but works on linux which is like learning a new thing from scratch, not that i cant, but will take up a lot of my time to reach to a level where a normal VoIP user on linux is currently compared to a windows user like me. in this part of the world, there r many guys selling Voip etc but all they know is of voipswitch as windows is what ppl know in this region. mayb i might be the only one from the whole region mostly using and promoting open source softwares like pfsense, opensbc and many such stuff and also might be the only one who knows there r things like asterisk and freeswitch. i searched like 75 outlets to get asterisk cards and all i got for an answer is, what is asterisk. living in this region and being able to test and try open source softwares is not easy coz never 2 guys meet and end up talking something about linux, pbx, softswitches etc coz no1 has ever heard of such things other than the stuff whats running in the economy. Brian West-3 wrote: > > > On Mar 30, 2009, at 9:33 AM, xbipin wrote: >> can any1 tell me where can i find a live cd image with the basic >> stuff to run >> FS and FS with all it tools installed and WITH A GUI, something like >> a pbx >> in a flash iso image so windows users like me find it easier to get >> testing >> with FS as the support for windows SIP proxy or any SIP related tool >> for >> windows platform is just about nil so i realized FS on windows also >> wont >> make much sense coz the rest of the developers etc use linux for FS >> and if i >> simply keep waiting for FS to actually do something productive on >> windows >> platform then it might take long or forever. > > Well this is a tall order... You know people ask for it.. or shall I > say demand it... but nobody really steps up to help out at all on the > GUI requests. > > FreeSWITCH on windows is equally capable minus a couple of things like > TLS since nobody will actually DO the work required to make it > happen. This isn't a buffet where you pull up and demand things be > one way or the other... this is a community where you start helping. > I would love to see more helping and less demanding! > >> if any i can provide me a live CD image with just enough tools to >> run FS to >> its fullest coz till date i have been only using Voipswitch and its >> time i >> need to implement TLS or any such type of encryption to reach new >> markets. > > > You could follow the linux how to and install CentOS and be done > already. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/live-iso-image-with-freeswitch-tp22784622p22786890.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Mar 30 09:26:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Mar 2009 09:26:38 -0700 Subject: [Freeswitch-users] [Remote SIP client] Couple of questions In-Reply-To: <7.0.1.0.2.20090325104634.02701c88@fredshack.com> References: <7.0.1.0.2.20090325104634.02701c88@fredshack.com> Message-ID: <87f2f3b90903300926n7f387eb2rc17a04463fafa4c9@mail.gmail.com> Just following up... did you get these questions ironed out? -MC On Wed, Mar 25, 2009 at 2:51 AM, Gilles wrote: > Hello, > > I have a couple of questions related to having SIP users connecting > from the Net to a Freeswitch server through NAT routers on both ends: > > 1. How must I configure routers on both ends? I understand that I > need to route incoming TCP/UDP 5080 into the Freeswitch server, but > what about the other router? I guess I also need to route this port > to let the SIP phone ring, but what about data (RTP/RTCP)? > > 2. The Freeswitch server is connected to the POTS with either an > OpenVox PCI card or a Linksys 3102 box: When a call is made between > the POTS and a remote SIP phone (ie. out there on the Net, not on the > same LAN as the Freeswitch server), is there a way for data to flow > directly from the POTS to the remote SIP client instead of through > the Freeswitch server? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sicfslist at gmail.com Mon Mar 30 09:29:23 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 30 Mar 2009 11:29:23 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22786890.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> Message-ID: <35b355e90903300929l17b0ac5ax67696d77b4fac4a7@mail.gmail.com> Here is a really good start --> http://wiki.freeswitch.org/wiki/SBC_Setup Overall I think you're going to just have to install Linux on a box and get after it. It may be more painful in the short term ... but in the long term your life will be much better. The devs / community here are pretty incredible with their desire and efforts to help everyone (and all platforms) ... but in all reality it's a huge task. If you stick with Linux / FS everything will just work and there is a tremendous amount of resources on the web. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/9aa7c85a/attachment.html From msc at freeswitch.org Mon Mar 30 09:33:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Mar 2009 09:33:02 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22786890.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> Message-ID: <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> > living in this region and being able to test and try open source softwares > is not easy coz never 2 guys meet and end up talking something about linux, > pbx, softswitches etc coz no1 has ever heard of such things other than the > stuff whats running in the economy. > Out of curiosity, what part of the world are you in? -MC From bipin at xbipin.com Mon Mar 30 10:26:25 2009 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 30 Mar 2009 21:26:25 +0400 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> Message-ID: <49D100C1.8010408@xbipin.com> hi, i currently live in a country called UAE - united arab emirates and a city called Dubai. Regards, Bipin -------- Original Message -------- Subject: Re: [Freeswitch-users] live iso image with freeswitch From: Michael Collins To: freeswitch-users at lists.freeswitch.org Date: Monday, March 30, 2009 8:33:02 PM >> living in this region and being able to test and try open source softwares >> is not easy coz never 2 guys meet and end up talking something about linux, >> pbx, softswitches etc coz no1 has ever heard of such things other than the >> stuff whats running in the economy. >> > > Out of curiosity, what part of the world are you in? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > __________ NOD32 3975 (20090330) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > > > From brian at freeswitch.org Mon Mar 30 10:30:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 12:30:27 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <49D100C1.8010408@xbipin.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> Message-ID: <2819EE9D-BFEA-4C0E-91BE-2A698E28CA9E@freeswitch.org> That is one interesting city.... I wouldn't mind paying a visit but its a bit rich for my blood! ;) On Mar 30, 2009, at 12:26 PM, Bipin Patel wrote: > hi, > > i currently live in a country called UAE - united arab emirates and a > city called Dubai. > > > Regards, > Bipin Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/674e1b3a/attachment.html From msc at freeswitch.org Mon Mar 30 10:36:40 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Mar 2009 10:36:40 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <49D100C1.8010408@xbipin.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> Message-ID: <87f2f3b90903301036u1c0d1c80g413db346d7cc2eef@mail.gmail.com> On Mon, Mar 30, 2009 at 10:26 AM, Bipin Patel wrote: > hi, > > i currently live in a country called UAE - united arab emirates and a > city called Dubai. > Hehe, Dubai is quite a popular place - even a number of us ignorant Americans have heard of it! :) We would love to see FreeSWITCH become more popular in Dubai since it is such an important business hub in the Arab world. Please keep checking back for updates on the subject of live CDs or ISO install images. They'll be ready sooner or later, hopefully sooner. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/ee6befdb/attachment.html From kristian.kielhofner at gmail.com Mon Mar 30 10:47:36 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 30 Mar 2009 13:47:36 -0400 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> Message-ID: <2d9149cd0903301047l65f654a2wd057427191520626@mail.gmail.com> The AstLinux ISO with FreeSWITCH is a live cd. http://mirror.astlinux.org/freeswitch/ This one is a little old but I could easily compile a new one... 2009/3/30 Brian West : > > On Mar 30, 2009, at 9:33 AM, xbipin wrote: > > can any1 tell me where can i find a live cd image with the basic stuff to > run > FS and FS with all it tools installed and WITH A GUI, something like a pbx > in a flash iso image so windows users like me find it easier to get testing > with FS as the support for windows SIP proxy or any SIP related tool for > windows platform is just about nil so i realized FS on windows also wont > make much sense coz the rest of the developers etc use linux for FS and if i > simply keep waiting for FS to actually do something productive on windows > platform then it might take long or forever. > > Well this is a tall order... You know people ask for it.. or shall I say > demand it... but nobody really steps up to help out at all on the GUI > requests. > FreeSWITCH on windows is equally capable minus a couple of things like TLS > since nobody will actually DO the work required to make it happen. ?This > isn't a buffet where you pull up and demand things be one way or the > other... this is a community where you start helping. ?I would love to see > more helping and less demanding! > > if any i can provide me a live CD image with just enough tools to run FS to > its fullest coz till date i have been only using Voipswitch and its time i > need to implement TLS or any such type of encryption to reach new markets. > > You could follow the linux how to and install CentOS and be done already. > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From gmaruzz at celliax.org Mon Mar 30 10:58:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 19:58:36 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <87f2f3b90903301036u1c0d1c80g413db346d7cc2eef@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <87f2f3b90903301036u1c0d1c80g413db346d7cc2eef@mail.gmail.com> Message-ID: <7b197bef0903301058m63f5c3afr74dd6b2c07f88a41@mail.gmail.com> Hi all, What I would like to stress is: 1) FreeSwitch is working on Windows, natively, without hacks 2) This is a huge advantage for a free software that want to be *really* popular (eg: be capable of running on an already working office machine, without dedicated hardware/expertise) 3) This is very important for people that are not "hard core", but just enthusiast, or just "wannabe". Why they have to go for a proprietary solution, maybe cracked? 4) This is very important for people/situation that just cannot afford another nmachine, or to dedicate a machine 5) Freeswitch is tested on Windows, albeit less than on *nix 6) This gap will be closing as the curve of adoption go further up 7) The efforts toward a GUI are proceeding, and in a multiplatform way, so they'll be working on Windows too I'm a *nix guy like you all, but let's bring closer to us, and us closer to, the vast majority of people/situations in the world. Especially making mind at the fact that the big effort, building FS multiplatform since the beginning, has been YET done :-) ! Ah, 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 2009/3/30 Michael Collins : > On Mon, Mar 30, 2009 at 10:26 AM, Bipin Patel wrote: >> >> hi, >> >> i currently live in a country called UAE - united arab emirates and a >> city called Dubai. > > Hehe, Dubai is quite a popular place - even a number of us ignorant > Americans have heard of it! :) We would love to see FreeSWITCH become more > popular in Dubai since it is such an important business hub in the Arab > world. Please keep checking back for updates on the subject of live CDs or > ISO install images. They'll be ready sooner or later, hopefully sooner. :) > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From gmaruzz at celliax.org Mon Mar 30 10:59:18 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 19:59:18 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <2d9149cd0903301047l65f654a2wd057427191520626@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <2d9149cd0903301047l65f654a2wd057427191520626@mail.gmail.com> Message-ID: <7b197bef0903301059i2683cf82xf078e9331e287d90@mail.gmail.com> Yes Kristian, please! On Mon, Mar 30, 2009 at 7:47 PM, Kristian Kielhofner wrote: > The AstLinux ISO with FreeSWITCH is a live cd. > > http://mirror.astlinux.org/freeswitch/ > > This one is a little old but I could easily compile a new one... > > 2009/3/30 Brian West : >> >> On Mar 30, 2009, at 9:33 AM, xbipin wrote: >> >> can any1 tell me where can i find a live cd image with the basic stuff to >> run >> FS and FS with all it tools installed and WITH A GUI, something like a pbx >> in a flash iso image so windows users like me find it easier to get testing >> with FS as the support for windows SIP proxy or any SIP related tool for >> windows platform is just about nil so i realized FS on windows also wont >> make much sense coz the rest of the developers etc use linux for FS and if i >> simply keep waiting for FS to actually do something productive on windows >> platform then it might take long or forever. >> >> Well this is a tall order... You know people ask for it.. or shall I say >> demand it... but nobody really steps up to help out at all on the GUI >> requests. >> FreeSWITCH on windows is equally capable minus a couple of things like TLS >> since nobody will actually DO the work required to make it happen. ?This >> isn't a buffet where you pull up and demand things be one way or the >> other... this is a community where you start helping. ?I would love to see >> more helping and less demanding! >> >> if any i can provide me a live CD image with just enough tools to run FS to >> its fullest coz till date i have been only using Voipswitch and its time i >> need to implement TLS or any such type of encryption to reach new markets. >> >> You could follow the linux how to and install CentOS and be done already. >> Brian West >> brian at freeswitch.org >> -- Meet us a ClueCon! ?http://www.cluecon.com >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mszlazak at aol.com Mon Mar 30 11:18:43 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 30 Mar 2009 14:18:43 -0400 Subject: [Freeswitch-users] Help scripting tone_detect. Message-ID: <8CB7F7B181E2057-1524-13D6@WEBMAIL-MZ40.sysops.aol.com> Except for fields in a dial plan extension, I can't get tone_detect (or stop_tone_detect), http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_stop_tone_detect to work in JavaScript. If there is one, what's the correct syntax for scripting these? Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/d3d6f2fa/attachment.html From gkuri at ieee.org Mon Mar 30 11:31:16 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Mon, 30 Mar 2009 11:31:16 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <49D100C1.8010408@xbipin.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> Message-ID: <49D10FF4.7090501@ieee.org> eh, how are poeple doing VoIP over there, given use of it outside the UAE is "officially" outlawed by the TRA ? I've heard Etisalat is pretty strict with making sure it's blocked going outside the country via a L7 packet inspection device to drop SIP. ~Gabe Bipin Patel wrote: > hi, > > i currently live in a country called UAE - united arab emirates and a > city called Dubai. > > > Regards, > Bipin > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] live iso image with freeswitch > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Date: Monday, March 30, 2009 8:33:02 PM > >>> living in this region and being able to test and try open source softwares >>> is not easy coz never 2 guys meet and end up talking something about linux, >>> pbx, softswitches etc coz no1 has ever heard of such things other than the >>> stuff whats running in the economy. >>> >> Out of curiosity, what part of the world are you in? >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> __________ NOD32 3975 (20090330) Information __________ >> >> This message was checked by NOD32 antivirus system. >> http://www.eset.com >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Mar 30 11:35:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 13:35:20 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <49D10FF4.7090501@ieee.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> Message-ID: <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> Really hard to inspect packets when they run on port 443 and are encrypted :P /b On Mar 30, 2009, at 1:31 PM, Gabriel Kuri wrote: > eh, how are poeple doing VoIP over there, given use of it outside the > UAE is "officially" outlawed by the TRA ? I've heard Etisalat is > pretty > strict with making sure it's blocked going outside the country via a > L7 > packet inspection device to drop SIP. > > ~Gabe Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/9978d78e/attachment.html From telles-listas at devel-it.com.br Mon Mar 30 11:35:50 2009 From: telles-listas at devel-it.com.br (Rodrigo P. Telles) Date: Mon, 30 Mar 2009 15:35:50 -0300 Subject: [Freeswitch-users] Response/reply match Nated endpoints Message-ID: <49D11106.4040409@devel-it.com.br> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/0fcaf2d0/attachment.html From kristian.kielhofner at gmail.com Mon Mar 30 11:41:30 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 30 Mar 2009 14:41:30 -0400 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> Message-ID: <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> Are you saying they should configure SIP TLS to run on port 443? :) 2009/3/30 Brian West : > Really hard to inspect packets when they run on port 443 and are encrypted > :P > /b -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From gkuri at ieee.org Mon Mar 30 11:48:16 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Mon, 30 Mar 2009 11:48:16 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> Message-ID: <49D113F0.7030204@ieee.org> yeah, obviously they'd have to enable TLS, but my question is more in terms of trying to use the mainstream providers outside the country, particularly for call termination/origination, which don't support TLS. ~Gabe Brian West wrote: > Really hard to inspect packets when they run on port 443 and are > encrypted :P > > /b > > On Mar 30, 2009, at 1:31 PM, Gabriel Kuri wrote: > >> eh, how are poeple doing VoIP over there, given use of it outside the >> UAE is "officially" outlawed by the TRA ? I've heard Etisalat is pretty >> strict with making sure it's blocked going outside the country via a L7 >> packet inspection device to drop SIP. >> >> ~Gabe > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Mon Mar 30 11:47:23 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 20:47:23 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> Message-ID: <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> no one would do that! On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner wrote: > Are you saying they should configure SIP TLS to run on port 443? :) > > 2009/3/30 Brian West : >> Really hard to inspect packets when they run on port 443 and are encrypted >> :P >> /b > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Mon Mar 30 11:48:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 13:48:08 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> Message-ID: <0E217C73-B2CB-40F4-9CE6-6BACC04D2E02@freeswitch.org> Even better run OpenVPN on port 53 :P /b On Mar 30, 2009, at 1:41 PM, Kristian Kielhofner wrote: > Are you saying they should configure SIP TLS to run on port 443? :) > > 2009/3/30 Brian West : >> Really hard to inspect packets when they run on port 443 and are >> encrypted >> :P >> /b Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/3090fba9/attachment.html From brian at freeswitch.org Mon Mar 30 11:49:14 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 13:49:14 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> Message-ID: You sure could to get around some oppression! :P /b On Mar 30, 2009, at 1:47 PM, Giovanni Maruzzelli wrote: > no one would do that! > > > On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner > wrote: >> Are you saying they should configure SIP TLS to run on port 443? :) >> >> 2009/3/30 Brian West : >>> Really hard to inspect packets when they run on port 443 and are >>> encrypted >>> :P >>> /b Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com From gmaruzz at celliax.org Mon Mar 30 11:51:43 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 20:51:43 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> Message-ID: <7b197bef0903301151lb97a157ua11bff974708213b@mail.gmail.com> ;-) On Mon, Mar 30, 2009 at 8:49 PM, Brian West wrote: > You sure could to get around some oppression! :P > > /b > > On Mar 30, 2009, at 1:47 PM, Giovanni Maruzzelli wrote: > >> no one would do that! >> >> >> On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner >> wrote: >>> Are you saying they should configure SIP TLS to run on port 443? :) >>> >>> 2009/3/30 Brian West : >>>> Really hard to inspect packets when they run on port 443 and are >>>> encrypted >>>> :P >>>> /b > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From can_man at gmx.de Mon Mar 30 13:33:53 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Mon, 30 Mar 2009 22:33:53 +0200 Subject: [Freeswitch-users] FS - MjSip no voice In-Reply-To: <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> References: <20090329220921.227090@gmx.net> <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> Message-ID: <20090330203353.59630@gmx.net> Hallo, thank you for your answer Anthony. > > starting at line 192 you seem to be sending yourself a notify, not sure > how you did that. That is indeed strange, I have looked at the MjSip code but haven't found the cause yet. > you are not by any chance trying to call a registered endpoint using the > FS > ip together with @ are you? > say you fs box is 1.2.3.4 and the phone is registered as 1000 > > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you > would > use sofia/internal/1000%1.2.3.4 > The % tells it to resolve the domain as a locally hosted domain and > translate it to the registered contact instead of using dns. > For testing I at the moment send the incoming call to the voicemail of user 1000 with this code: return '''\n'''\ '''\n'''\ '''
\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''
\n'''\ '''''' % (didNumber, didNumber, id) Works fine with a normal SIP client. I have captured more output with debug enabled and have also captured the SIP messages originating from MjSip. FS: http://pastebin.freeswitch.org/8045 MjSip: http://pastebin.freeswitch.org/8046 Thank you very much for your help. Best wishes, Phil > > > On Sun, Mar 29, 2009 at 5:09 PM, wrote: > > > Hello everyone, > > > > I am trying to get FS working with the MjSip Java Sip-stack, the > SipToSis > > source and the normal one. Everything works well within my own network > and > > when using x-lite, but when it comes to making calls from MjSip to an > > outside FS server I don't hear any voice - seems to be a NAT problem or > some > > kind of other MjSip problem. Registration works fine though and SIP > messages > > get through ok, but non of the UDP RTP ones. Would be great if someone > could > > advice me on how to do the setup correctly. > > > > The whole FS trace can be found here: > http://pastebin.freeswitch.org/8029 > > > > The settings for MjSip are: > > > > "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090", > > "transport_protocols=udp tcp","from_url=", > > > > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > > > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > > > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > > "bin_rat=rat","bin_vic=vic" > > > > > > Thank you very much. > > Best wishes, > > Phil > > > > -- > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From anthony.minessale at gmail.com Mon Mar 30 14:20:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 16:20:19 -0500 Subject: [Freeswitch-users] FS - MjSip no voice In-Reply-To: <20090330203353.59630@gmx.net> References: <20090329220921.227090@gmx.net> <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> <20090330203353.59630@gmx.net> Message-ID: <191c3a030903301420l78f848ft281430e4d67daa0e@mail.gmail.com> maybe that phone does not support early media try adding the answer application to your dialplan On Mon, Mar 30, 2009 at 3:33 PM, wrote: > Hallo, > > thank you for your answer Anthony. > > > > > starting at line 192 you seem to be sending yourself a notify, not sure > > how you did that. > > That is indeed strange, I have looked at the MjSip code but haven't found > the cause yet. > > > you are not by any chance trying to call a registered endpoint using the > > FS > > ip together with @ are you? > > say you fs box is 1.2.3.4 and the phone is registered as 1000 > > > > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you > > would > > use sofia/internal/1000%1.2.3.4 > > The % tells it to resolve the domain as a locally hosted domain and > > translate it to the registered contact instead of using dns. > > > > For testing I at the moment send the incoming call to the voicemail of user > 1000 with this code: > > return '''\n'''\ > '''\n'''\ > '''
\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''
\n'''\ > '''''' % (didNumber, didNumber, id) > > > Works fine with a normal SIP client. > I have captured more output with debug enabled and have also captured the > SIP messages originating from MjSip. > > FS: http://pastebin.freeswitch.org/8045 > MjSip: http://pastebin.freeswitch.org/8046 > > Thank you very much for your help. > Best wishes, > Phil > > > > > > > On Sun, Mar 29, 2009 at 5:09 PM, wrote: > > > > > Hello everyone, > > > > > > I am trying to get FS working with the MjSip Java Sip-stack, the > > SipToSis > > > source and the normal one. Everything works well within my own network > > and > > > when using x-lite, but when it comes to making calls from MjSip to an > > > outside FS server I don't hear any voice - seems to be a NAT problem or > > some > > > kind of other MjSip problem. Registration works fine though and SIP > > messages > > > get through ok, but non of the UDP RTP ones. Would be great if someone > > could > > > advice me on how to do the setup correctly. > > > > > > The whole FS trace can be found here: > > http://pastebin.freeswitch.org/8029 > > > > > > The settings for MjSip are: > > > > > > "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090", > > > "transport_protocols=udp tcp","from_url= >", > > > > > > > > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > > > > > > > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > > > > > > > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > > > "bin_rat=rat","bin_vic=vic" > > > > > > > > > Thank you very much. > > > Best wishes, > > > Phil > > > > > > -- > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > -- > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/db3ed669/attachment.html From keithl at voxtelecom.co.za Mon Mar 30 14:56:57 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Mon, 30 Mar 2009 23:56:57 +0200 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. Message-ID: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> Hi, I have an application where my Javascript hanguphook code calculates a value (e.g. the cost of the call which can only be calculated post hangup) and I need to have that value appear as a field in the cdrs. As the 'session' object is no longer available for javascript logic post hangup, I can't figure out how to 'set' a user variable post hangup, such that it can be written to the cdr when the state changes from CS_HANGUP -> CS_REPORTING. Maybe it's just not possible......? It would be a pity to have to resort to writing out cdrs from the javascript itself and duplicating what fs does so well already. Any advice / suggestions would be appreciated. Best Regards Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/f80d0c7f/attachment-0001.html From msc at freeswitch.org Mon Mar 30 15:08:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Mar 2009 15:08:41 -0700 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> Message-ID: <87f2f3b90903301508v5f6a5200qd918aa1fa5f280a1@mail.gmail.com> Are logging both a- and b-legs? Just curious what your setup is. -MC 2009/3/30 Keith Laaks > Hi, > > > > I have an application where my Javascript hanguphook code calculates a > value (e.g. the cost of the call which can only be calculated post hangup) > and I need to have that value appear as a field in the cdrs. > > > > As the ?session? object is no longer available for javascript logic post > hangup, I can?t figure out how to ?set? a user variable post hangup, such > that it can be written to the cdr when the state changes from CS_HANGUP -> > CS_REPORTING. Maybe it?s just not possible??? It would be a pity to have to > resort to writing out cdrs from the javascript itself and duplicating what > fs does so well already. > > > > Any advice / suggestions would be appreciated. > > > > Best Regards > > > > Keith > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/520305c7/attachment.html From anthony.minessale at gmail.com Mon Mar 30 15:12:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 17:12:58 -0500 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> Message-ID: <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> in your script called via api_hangup_hook: var env = request.dumpENV("text"); consoleLog("info", env); all those vars are there for you, you can get the individually with var hval = request.getHeader("some_header"); 2009/3/30 Keith Laaks > Hi, > > > > I have an application where my Javascript hanguphook code calculates a > value (e.g. the cost of the call which can only be calculated post hangup) > and I need to have that value appear as a field in the cdrs. > > > > As the ?session? object is no longer available for javascript logic post > hangup, I can?t figure out how to ?set? a user variable post hangup, such > that it can be written to the cdr when the state changes from CS_HANGUP -> > CS_REPORTING. Maybe it?s just not possible??? It would be a pity to have to > resort to writing out cdrs from the javascript itself and duplicating what > fs does so well already. > > > > Any advice / suggestions would be appreciated. > > > > Best Regards > > > > Keith > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/282a43bc/attachment.html From dule.maillist at gmail.com Mon Mar 30 16:44:05 2009 From: dule.maillist at gmail.com (Dan Le) Date: Mon, 30 Mar 2009 19:44:05 -0400 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <49CC8DFE.3050104@gmx.net> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> Message-ID: <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> I get similar behavior as Peter when trying to enter a locked conference. If I am just dialing from a phone to a conference (on a dialplan), it will properly lock me out. But if I do an originate command (originate sofia/internal/1001 &conference(3000)), it will drop me into the conference, even though it is suppose to be locked. I am using the released 1.0.3 tag. On Fri, Mar 27, 2009 at 4:27 AM, Peter P GMX wrote: > It's defined via XML-Curl, and manual dialling and transfering do > trigger the same xml-curl request. This means that this conference > number is not defined in the any xml conf file. > If I transfer a call (without PIN) and then manually dial with another > phone into this conf with PIN, both calls are in the same conference. > > I have SVN rev 12796. > > > Best regards > Peter > > > Michael Collins schrieb: > > On Thu, Mar 26, 2009 at 4:09 PM, Peter P GMX > wrote: > > > >> Hello Michael, > >> > >> I tried this, but received the same behaviour. It does not ask for the > >> defined PIN. > >> > > > > Just curious - where do you define the PIN for this conference? > > -MC > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/fd360936/attachment.html From brian at freeswitch.org Mon Mar 30 16:54:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 18:54:08 -0500 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> Message-ID: <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> Update again to svn trunk... btw 1.0.4 pre3 is out on files.freeswitch.org /b On Mar 30, 2009, at 6:44 PM, Dan Le wrote: > I get similar behavior as Peter when trying to enter a locked > conference. > > If I am just dialing from a phone to a conference (on a dialplan), > it will properly lock me out. But if I do an originate command > (originate sofia/internal/1001 &conference(3000)), it will drop me > into the conference, even though it is suppose to be locked. > > I am using the released 1.0.3 tag. > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/be7c929e/attachment.html From bipin at xbipin.com Mon Mar 30 21:55:28 2009 From: bipin at xbipin.com (xbipin) Date: Mon, 30 Mar 2009 21:55:28 -0700 (PDT) Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> Message-ID: <22798341.post@talk.nabble.com> hi, i think this thread has got a bit too much of attention but in all directions, bytheway sorry for that double posting but i think its something to do with nabble as i get a mail saying ur post is rejected so i resend and then i find both the posts get accepted. UAE is very strict in terms of allowing VoIP, they block skype and any sort of voip available but come to see it, they use all the latest gadgets to block advanced stuff but its basic stuff that bypasses it thats y guys like me end up selling voip and any1 dealing in UAE will only tell u one thing, its a million dollar market for voip. simple technic to bypass is, use any port other than the standard SIP ports and in the SIP packet, if the header says SIP or something like that, change it to SI_P or S_IP or whatever u like as the header and ur through the block. known methods to bypass r openvpn but they find out and simply block the port, XOR or CBCOM as said by grandstream, voipswitch's voiptunnel and just about anything that doesnt say anything about SIP or RTP in the packet header or description or whatever it may be. bytheway that astlinux is for embedded systems etc right? i have a ALIX board already running pfsense, im looking for a normal linux image for a workstation but just enough tools to run freeswitch, have the gnome or K desktop environment and later on when the web GUI is developed then simply the web GUI. -- View this message in context: http://www.nabble.com/live-iso-image-with-freeswitch-tp22784622p22798341.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From moizchinoy at gmail.com Mon Mar 30 23:00:05 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Tue, 31 Mar 2009 10:00:05 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <191c3a030903300607m55d7dcd8u78e8ea9fc7b491cb@mail.gmail.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> <29b888f80903300056t51828330s39bbf662fe2cd8c0@mail.gmail.com> <191c3a030903300607m55d7dcd8u78e8ea9fc7b491cb@mail.gmail.com> Message-ID: <29b888f80903302300q1eb097cey2178fded546984af@mail.gmail.com> This I/O error occurred once. And I am able to communicate google talk to SIP softphone through FS. Still does not know why this error happened as I did not change anything! 2009/3/30 Anthony Minessale : > libdingaling.c:1545 xmpp_connect() io error 2 7 > > It means there was an I/O error trying to connect > > it's probably because iksemel does not have srv support, if you are trying > to connect to gtalk > you have to manually specify the server as talk.google.com, the default > client.xml is self-explanatory for > connecting to gtalk and it's on the wiki too.. > > > On Mon, Mar 30, 2009 at 2:56 AM, Moiz Chinoy wrote: >> >> Hi, >> >> If there is no build for the tls with freeswitch on windows, can you >> please guide how to test gtalk integration on windows. >> >> I have successfully compiled the tls with iksemel, there were couple >> of errors but I managed to compile it. >> >> Now, I am also getting the error libdingaling.c:1545 xmpp_connect() io >> error 2 7 >> >> What is this error about? >> >> There is a similar post on the mailing list regarding the above IO >> error but can't find the reply for it. >> >> On Sat, Mar 28, 2009 at 8:03 PM, Michael Jerris wrote: >> > please see my previous response in this thread. >> > >> > MIke >> > >> > On Mar 28, 2009, at 5:09 AM, Moiz Chinoy wrote: >> > >> >> I am trying it on windows. >> >> Where to get gnutls-devel for windows? >> >> >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Regards, >> Moiz Chinoy. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Moiz Chinoy. From gmaruzz at celliax.org Tue Mar 31 00:26:57 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 31 Mar 2009 09:26:57 +0200 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To Message-ID: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS Microsoft Windows Download 118MB HD: http://www.celliax.org/final.avi 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 From jason at jasonjgw.net Tue Mar 31 00:28:58 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 31 Mar 2009 18:28:58 +1100 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> Message-ID: <20090331072858.GA15463@jdc.jasonjgw.net> Brian West wrote: > This isn't a buffet where you pull up and demand things be one way or the > other... this is a community where you start helping. I would love to > see more helping and less demanding! So would I. I regularly scan the mailing list looking for questions to answer, but many of them relate to scenarios of which I have had no experience, or features that I haven't had any reason to use. I am worried that the questions are answered by the same people much of the time, which in the long run will be bad for the project as the community grows, i.e., more technical support work for the same people (who are mostly the core developers as well) is not a sustainable proposition. How do other large projects handle this? Does anyone have any positive suggestions to offer that would encourage more contributions? From bipin at xbipin.com Tue Mar 31 01:22:16 2009 From: bipin at xbipin.com (xbipin) Date: Tue, 31 Mar 2009 01:22:16 -0700 (PDT) Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> Message-ID: <22800505.post@talk.nabble.com> hi, is it just audio or is it that im having broken codecs so cant view any video? Regards, Bipin Giovanni Maruzzelli-3 wrote: > > Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS > Microsoft Windows > Download 118MB HD: http://www.celliax.org/final.avi > > 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p22800505.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From regs at kinetix.gr Tue Mar 31 02:03:10 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 31 Mar 2009 12:03:10 +0300 Subject: [Freeswitch-users] killall -HUP freeswitch Message-ID: <49D1DC4E.7070002@kinetix.gr> Hi, I am doing a "killall -HUP freeswitch" in order to achieve cdr-csv log rotation and the process gets killed instead. A few days ago it worked fine. I also noticed that a few days ago when I hit Ctrl+C freeswitch did not exit. Now, when I do this the process gets terminated immediately. What could be the cause of all this? I am using the svn 12548 trunk. I did not do any recompiling of FS. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From regs at kinetix.gr Tue Mar 31 02:10:23 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 31 Mar 2009 12:10:23 +0300 Subject: [Freeswitch-users] killall -HUP freeswitch In-Reply-To: <49D1DC4E.7070002@kinetix.gr> References: <49D1DC4E.7070002@kinetix.gr> Message-ID: <49D1DDFF.2080402@kinetix.gr> I found out the cause : When the mod_opal module is loaded the FS process gets killed with a kill -HUP. I thought it would be good for everyone to know. Apostolos Pantsiopoulos wrote: > Hi, > > I am doing a "killall -HUP freeswitch" in order to achieve > cdr-csv log rotation and the process gets killed instead. A few > days ago it worked fine. I also noticed that a few days ago when I > hit Ctrl+C freeswitch did not exit. Now, when I do this the process gets > terminated immediately. What could be the cause of all this? I am using > the svn 12548 trunk. I did not do any recompiling of FS. > > > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From solko at gcdf.pl Tue Mar 31 03:05:57 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 31 Mar 2009 12:05:57 +0200 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <22800505.post@talk.nabble.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> Message-ID: <49D1EB05.7090804@gcdf.pl> xbipin pisze: > hi, > > is it just audio or is it that im having broken codecs so cant view any > video? > There are both, video and audio. Mplayer dump Otwieram dekoder video: [ffmpeg] FFmpeg's libavcodec codec family Wybrany kodek video: [ffcamtasia] vfm: ffmpeg (TechSmith Camtasia Screen Codec (native)) ========================================================================== ========================================================================== Otwieram dekoder audio: [pcm] Uncompressed PCM audio decoder AUDIO: 22050 Hz, 1 ch, s16le, 352.8 kbit/100.00% (ratio: 44100->44100) Wybrany kodek audio: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== > Regards, > Bipin > > > > Giovanni Maruzzelli-3 wrote: >> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS >> Microsoft Windows >> Download 118MB HD: http://www.celliax.org/final.avi >> >> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From keithl at voxtelecom.co.za Tue Mar 31 04:25:12 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Tue, 31 Mar 2009 13:25:12 +0200 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> Message-ID: <1B99233662E2104983E3550185D3ED73498352@xena.internal.datapro.co.za> Hi, Here is what I am trying to accomplish: //--- completecall.js trigged via api_hangup_hook ---- use("CURL"); const loglevel='notice'; var uuid = request.getHeader("Core-UUID"); var billmsec = request.getHeader("billmsec"); var urlrequest = "UUID=" + uuid + "&billmsec=" + billmsec; function reply_callback(string, arg) { string = string.substring(string.search("API")+3); string = string.substring(0,string.search("<")); var splits = string.split("~"); var i = 0; var length = splits.length; for (i=0; i < length; i++) { var fv=splits[i].split("="); console_log(loglevel, "setting: " + fv[0] + " = " + fv[1] + "\n"); //session.setVariable(fv[0],fv[1]); // Cant use set here as the session is dead by now - the call has been terminated } return true; } var curl = new CURL(); console_log(loglevel,"-- completecall.js ->" + url + "?" + urlrequest + "\n"); var env = request.dumpENV("text"); // debug console_log("ENV:\n", env + "\n"); // debug curl.run("POST", "http://127.0.0.1:10502/rate", urlrequest, reply_callback, "CBrate\n"); //returns string like APIcharge=10.20 exit(); So the trick is that I am accessing an external system via CURL where the call rate is calculated and returned (based on the call duration and uuid). Now I need to use a 'setVariable' to get it back as one of the parameters that the cdr module can write out for me. Note that in the above, the external system will return 'charge=value'. I need to set the variable 'charge' to the value in 'value'. Then using the config below, 'charge' can be written out as one of the cdr fields. Best Regards Keith From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: 31 March 2009 00:13 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Javascript, Hanguphooks,CDRs and User Variables. in your script called via api_hangup_hook: var env = request.dumpENV("text"); consoleLog("info", env); all those vars are there for you, you can get the individually with var hval = request.getHeader("some_header"); 2009/3/30 Keith Laaks Hi, I have an application where my Javascript hanguphook code calculates a value (e.g. the cost of the call which can only be calculated post hangup) and I need to have that value appear as a field in the cdrs. As the 'session' object is no longer available for javascript logic post hangup, I can't figure out how to 'set' a user variable post hangup, such that it can be written to the cdr when the state changes from CS_HANGUP -> CS_REPORTING. Maybe it's just not possible......? It would be a pity to have to resort to writing out cdrs from the javascript itself and duplicating what fs does so well already. Any advice / suggestions would be appreciated. Best Regards Keith _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/ff03685f/attachment-0001.html From anthony.minessale at gmail.com Tue Mar 31 05:59:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 07:59:34 -0500 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <1B99233662E2104983E3550185D3ED73498352@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> <1B99233662E2104983E3550185D3ED73498352@xena.internal.datapro.co.za> Message-ID: <191c3a030903310559g49211e66x3cbdc1dac088e671@mail.gmail.com> if you set the channel variable 'session_in_hangup_hook=true' early in the call, the session will be present in your script. 2009/3/31 Keith Laaks > Hi, > > > > Here is what I am trying to accomplish: > > > > //--- completecall.js trigged via api_hangup_hook ---- > > > > use("CURL"); > > > > const loglevel='notice'; > > var uuid = request.getHeader("Core-UUID"); > > var billmsec = request.getHeader("billmsec"); > > var urlrequest = "UUID=" + uuid + "&billmsec=" + billmsec; > > > > function reply_callback(string, arg) { > > string = string.substring(string.search("API")+3); > > string = string.substring(0,string.search("<")); > > var splits = string.split("~"); > > var i = 0; > > var length = splits.length; > > for (i=0; i < length; i++) { > > var fv=splits[i].split("="); > > console_log(loglevel, "setting: " + fv[0] + " = " + fv[1] + "\n"); > > //session.setVariable(fv[0],fv[1]); // Cant use set here as the session > is dead by now - the call has been terminated > > } > > return true; > > } > > > > var curl = new CURL(); > > console_log(loglevel,"-- completecall.js ->" + url + "?" + urlrequest + > "\n"); > > > > var env = request.dumpENV("text"); // debug > > console_log("ENV:\n", env + "\n"); // debug > > > > curl.run("POST", "http://127.0.0.1:10502/rate", urlrequest, > reply_callback, "CBrate\n"); > > //returns string like APIcharge=10.20 > > > > exit(); > > > > > > So the trick is that I am accessing an external system via CURL where the > call rate is calculated and returned (based on the call duration and uuid). > > Now I need to use a ?setVariable? to get it back as one of the parameters > that the cdr module can write out for me. > > Note that in the above, the external system will return ?charge=value?. > > I need to set the variable ?charge? to the value in ?value?. > > Then using the config below, ?charge? can be written out as one of the cdr > fields. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Best Regards > > > > Keith > > > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* 31 March 2009 00:13 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Javascript, Hanguphooks,CDRs and User > Variables. > > > > in your script called via api_hangup_hook: > > var env = request.dumpENV("text"); > > consoleLog("info", env); > > all those vars are there for you, you can get the individually with > var hval = request.getHeader("some_header"); > > 2009/3/30 Keith Laaks > > Hi, > > > > I have an application where my Javascript hanguphook code calculates a > value (e.g. the cost of the call which can only be calculated post hangup) > and I need to have that value appear as a field in the cdrs. > > > > As the ?session? object is no longer available for javascript logic post > hangup, I can?t figure out how to ?set? a user variable post hangup, such > that it can be written to the cdr when the state changes from CS_HANGUP -> > CS_REPORTING. Maybe it?s just not possible??? It would be a pity to have to > resort to writing out cdrs from the javascript itself and duplicating what > fs does so well already. > > > > Any advice / suggestions would be appreciated. > > > > Best Regards > > > > Keith > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/76a269c0/attachment.html From andy at fabulous4.co.uk Tue Mar 31 06:04:19 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 31 Mar 2009 14:04:19 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message Message-ID: <30BF7571B7FC4F029C643D0E479CE614@wsandy> Hi, I'm using freeswitch as a glorified answering machine. FS registers with a VOIP gateway and all calls into the gateway go through an ivr menu and are allowed to leave a message which gets recorded to a file. The FS box is behind a NAT firewall. Everything works fine except that intermittently, calls keep getting cut off after a number of seconds. I've attached a snapshot of the log at the point that the call gets cut off, can anyone suggest why this is happening or how I can prevent it? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/81df0c39/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: log-snippet.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/81df0c39/attachment-0001.txt From brian at freeswitch.org Tue Mar 31 06:26:41 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 08:26:41 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <30BF7571B7FC4F029C643D0E479CE614@wsandy> References: <30BF7571B7FC4F029C643D0E479CE614@wsandy> Message-ID: I'm going to guess you're not on SVN trunk? what rev are you on? /b On Mar 31, 2009, at 8:04 AM, Andy Ayers wrote: > Hi, > > I'm using freeswitch as a glorified answering machine. FS registers > with a VOIP gateway and all calls into the gateway go through an ivr > menu and are allowed to leave a message which gets recorded to a > file. The FS box is behind a NAT firewall. Everything works fine > except that intermittently, calls keep getting cut off after a > number of seconds. > > I've attached a snapshot of the log at the point that the call gets > cut off, can anyone suggest why this is happening or how I can > prevent it? > > Many thanks > Andy > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/0686c6e9/attachment.html From andy at fabulous4.co.uk Tue Mar 31 06:36:29 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 31 Mar 2009 14:36:29 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: <8B544F779B6948C9BF77DBE6D5599CFA@wsandy> Hi Brian, 1.03 Thanks Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:27 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message I'm going to guess you're not on SVN trunk? what rev are you on? /b On Mar 31, 2009, at 8:04 AM, Andy Ayers wrote: Hi, I'm using freeswitch as a glorified answering machine. FS registers with a VOIP gateway and all calls into the gateway go through an ivr menu and are allowed to leave a message which gets recorded to a file. The FS box is behind a NAT firewall. Everything works fine except that intermittently, calls keep getting cut off after a number of seconds. I've attached a snapshot of the log at the point that the call gets cut off, can anyone suggest why this is happening or how I can prevent it? Many thanks Andy _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/264b1809/attachment.html From brian at freeswitch.org Tue Mar 31 06:39:34 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 08:39:34 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <8B544F779B6948C9BF77DBE6D5599CFA@wsandy> References: <8B544F779B6948C9BF77DBE6D5599CFA@wsandy> Message-ID: Please try SVN trunk. /b On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote: > Hi Brian, > > 1.03 > > Thanks > Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/118180a9/attachment.html From can_man at gmx.de Tue Mar 31 07:06:48 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Tue, 31 Mar 2009 16:06:48 +0200 Subject: [Freeswitch-users] FS - MjSip no voice [SOLVED] SIP 200 / 183 problem In-Reply-To: <191c3a030903301420l78f848ft281430e4d67daa0e@mail.gmail.com> References: <20090329220921.227090@gmx.net> <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> <20090330203353.59630@gmx.net> <191c3a030903301420l78f848ft281430e4d67daa0e@mail.gmail.com> Message-ID: <20090331140648.227850@gmx.net> Hello, I have found the problem. FS on my local network sends "SIP/2.0 200 OK" after an invite and FS on the net through the external profil sends SIP/2.0 183 Session Progress. But MjSip doesn't know how to deal with 183, so it just ignores the message. For testing I have changed the 183 header to the 200 one and now it works. Thank you for your help and the quick response time. Best wishes, Phil >From FS on the net through the external profil: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 90.181.59.141:5090;rport=60315;branch=z9hG4bK256321;received=78.105.17.88 From: ;tag=z9hG4bK40977269 To: ;tag=vgg3Zja8pNQcg Call-ID: 507347917247 at 90.181.59.141 CSeq: 1 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12839M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 267 v=0 o=FreeSWITCH 1072777625698755085 8893522831081357051 IN IP4 90.181.59.141 s=FreeSWITCH c=IN IP4 91.121.59.148 t=0 0 m=audio 26722 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 >From FS in my local network: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.143:5060;rport=5060;branch=z9hG4bK423233;received=192.168.1.102 From: ;tag=z9hG4bK42598163 To: ;tag=Q0X494ZUNaKHH Call-ID: 961142687222 at 192.168.1.143 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12712M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 267 v=0 o=FreeSWITCH 5195745633884389954 8941954824002056485 IN IP4 192.168.1.143 s=FreeSWITCH c=IN IP4 192.168.1.143 t=0 0 m=audio 22680 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 > maybe that phone does not support early media > > try adding the answer application to your dialplan > > > On Mon, Mar 30, 2009 at 3:33 PM, wrote: > > > Hallo, > > > > thank you for your answer Anthony. > > > > > > > > starting at line 192 you seem to be sending yourself a notify, not > sure > > > how you did that. > > > > That is indeed strange, I have looked at the MjSip code but haven't > found > > the cause yet. > > > > > you are not by any chance trying to call a registered endpoint using > the > > > FS > > > ip together with @ are you? > > > say you fs box is 1.2.3.4 and the phone is registered as 1000 > > > > > > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you > > > would > > > use sofia/internal/1000%1.2.3.4 > > > The % tells it to resolve the domain as a locally hosted domain and > > > translate it to the registered contact instead of using dns. > > > > > > > For testing I at the moment send the incoming call to the voicemail of > user > > 1000 with this code: > > > > return '''\n'''\ > > '''\n'''\ > > '''
\n'''\ > > '''\n'''\ > > '''\n'''\ > > ''' expression="^(%s)$">\n'''\ > > '''\n'''\ > > '''\n'''\ > > '''\n'''\ > > '''\n'''\ > > '''
\n'''\ > > '''''' % (didNumber, didNumber, id) > > > > > > Works fine with a normal SIP client. > > I have captured more output with debug enabled and have also captured > the > > SIP messages originating from MjSip. > > > > FS: http://pastebin.freeswitch.org/8045 > > MjSip: http://pastebin.freeswitch.org/8046 > > > > Thank you very much for your help. > > Best wishes, > > Phil > > > > > > > > > > > On Sun, Mar 29, 2009 at 5:09 PM, wrote: > > > > > > > Hello everyone, > > > > > > > > I am trying to get FS working with the MjSip Java Sip-stack, the > > > SipToSis > > > > source and the normal one. Everything works well within my own > network > > > and > > > > when using x-lite, but when it comes to making calls from MjSip to > an > > > > outside FS server I don't hear any voice - seems to be a NAT problem > or > > > some > > > > kind of other MjSip problem. Registration works fine though and SIP > > > messages > > > > get through ok, but non of the UDP RTP ones. Would be great if > someone > > > could > > > > advice me on how to do the setup correctly. > > > > > > > > The whole FS trace can be found here: > > > http://pastebin.freeswitch.org/8029 > > > > > > > > The settings for MjSip are: > > > > > > > > "via_addr=91.101.58.142 (changed in the whole > trace)","host_port=5090", > > > > "transport_protocols=udp tcp","from_url= > >", > > > > > > > > > > > > > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > > > > > > > > > > > > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > > > > > > > > > > > > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > > > > "bin_rat=rat","bin_vic=vic" > > > > > > > > > > > > Thank you very much. > > > > Best wishes, > > > > Phil > > > > > > > > -- > > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > < > > > MSN%3Aanthony_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 > < > > > sip%3A888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > pstn:213-799-1400 > > > > -- > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01 From msc at freeswitch.org Tue Mar 31 07:13:01 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 31 Mar 2009 07:13:01 -0700 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <191c3a030903310559g49211e66x3cbdc1dac088e671@mail.gmail.com> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> <1B99233662E2104983E3550185D3ED73498352@xena.internal.datapro.co.za> <191c3a030903310559g49211e66x3cbdc1dac088e671@mail.gmail.com> Message-ID: <70D31887-92F2-498C-97D2-AE7797544E97@freeswitch.org> On Mar 31, 2009, at 5:59 AM, Anthony Minessale wrote: > if you set the channel variable 'session_in_hangup_hook=true' early > in the call, the session will be present in your script. > Very cool. I will get this chan var documented on the wiki right away. -MC From anthony.minessale at gmail.com Tue Mar 31 07:42:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 09:42:27 -0500 Subject: [Freeswitch-users] FS - MjSip no voice [SOLVED] SIP 200 / 183 problem In-Reply-To: <20090331140648.227850@gmx.net> References: <20090329220921.227090@gmx.net> <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> <20090330203353.59630@gmx.net> <191c3a030903301420l78f848ft281430e4d67daa0e@mail.gmail.com> <20090331140648.227850@gmx.net> Message-ID: <191c3a030903310742q15edc6acn8d2f0227e0e2347b@mail.gmail.com> like i said: > maybe that phone does not support early media > > try adding the answer application to your dialplan early media == 183 answer = 200 it depends on your dialplan in FS On Tue, Mar 31, 2009 at 9:06 AM, wrote: > Hello, > > I have found the problem. FS on my local network sends "SIP/2.0 200 OK" > after an invite and FS on the net through the external profil sends > SIP/2.0 183 Session Progress. But MjSip doesn't know how to deal with > 183, so it just ignores the message. For testing I have changed > the 183 header to the 200 one and now it works. > > Thank you for your help and the quick response time. > Best wishes, > Phil > > > >From FS on the net through the external profil: > > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP 90.181.59.141:5090 > ;rport=60315;branch=z9hG4bK256321;received=78.105.17.88 > From: ;tag=z9hG4bK40977269 > To: ;tag=vgg3Zja8pNQcg > Call-ID: 507347917247 at 90.181.59.141 > CSeq: 1 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12839M > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 267 > > v=0 > o=FreeSWITCH 1072777625698755085 8893522831081357051 IN IP4 90.181.59.141 > s=FreeSWITCH > c=IN IP4 91.121.59.148 > t=0 0 > m=audio 26722 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > > > >From FS in my local network: > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.143:5060 > ;rport=5060;branch=z9hG4bK423233;received=192.168.1.102 > From: > >;tag=z9hG4bK42598163 > To: >;tag=Q0X494ZUNaKHH > Call-ID: 961142687222 at 192.168.1.143 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12712M > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 267 > > v=0 > o=FreeSWITCH 5195745633884389954 8941954824002056485 IN IP4 192.168.1.143 > s=FreeSWITCH > c=IN IP4 192.168.1.143 > t=0 0 > m=audio 22680 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > > > > > maybe that phone does not support early media > > > > try adding the answer application to your dialplan > > > > > > On Mon, Mar 30, 2009 at 3:33 PM, wrote: > > > > > Hallo, > > > > > > thank you for your answer Anthony. > > > > > > > > > > > starting at line 192 you seem to be sending yourself a notify, not > > sure > > > > how you did that. > > > > > > That is indeed strange, I have looked at the MjSip code but haven't > > found > > > the cause yet. > > > > > > > you are not by any chance trying to call a registered endpoint using > > the > > > > FS > > > > ip together with @ are you? > > > > say you fs box is 1.2.3.4 and the phone is registered as 1000 > > > > > > > > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4you > > > > would > > > > use sofia/internal/1000%1.2.3.4 > > > > The % tells it to resolve the domain as a locally hosted domain and > > > > translate it to the registered contact instead of using dns. > > > > > > > > > > For testing I at the moment send the incoming call to the voicemail of > > user > > > 1000 with this code: > > > > > > return '''\n'''\ > > > '''\n'''\ > > > '''
\n'''\ > > > '''\n'''\ > > > '''\n'''\ > > > ''' > expression="^(%s)$">\n'''\ > > > '''\n'''\ > > > '''\n'''\ > > > '''\n'''\ > > > '''\n'''\ > > > '''
\n'''\ > > > '''''' % (didNumber, didNumber, id) > > > > > > > > > Works fine with a normal SIP client. > > > I have captured more output with debug enabled and have also captured > > the > > > SIP messages originating from MjSip. > > > > > > FS: http://pastebin.freeswitch.org/8045 > > > MjSip: http://pastebin.freeswitch.org/8046 > > > > > > Thank you very much for your help. > > > Best wishes, > > > Phil > > > > > > > > > > > > > > > On Sun, Mar 29, 2009 at 5:09 PM, wrote: > > > > > > > > > Hello everyone, > > > > > > > > > > I am trying to get FS working with the MjSip Java Sip-stack, the > > > > SipToSis > > > > > source and the normal one. Everything works well within my own > > network > > > > and > > > > > when using x-lite, but when it comes to making calls from MjSip to > > an > > > > > outside FS server I don't hear any voice - seems to be a NAT > problem > > or > > > > some > > > > > kind of other MjSip problem. Registration works fine though and SIP > > > > messages > > > > > get through ok, but non of the UDP RTP ones. Would be great if > > someone > > > > could > > > > > advice me on how to do the setup correctly. > > > > > > > > > > The whole FS trace can be found here: > > > > http://pastebin.freeswitch.org/8029 > > > > > > > > > > The settings for MjSip are: > > > > > > > > > > "via_addr=91.101.58.142 (changed in the whole > > trace)","host_port=5090", > > > > > "transport_protocols=udp tcp","from_url=< > sip:puli at 91.101.58.142:5090 > > > >", > > > > > > > > > > > > > > > > > > > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > > > > > > > > > > > > > > > > > > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > > > > > > > > > > > > > > > > > > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > > > > > "bin_rat=rat","bin_vic=vic" > > > > > > > > > > > > > > > Thank you very much. > > > > > Best wishes, > > > > > Phil > > > > > > > > > > -- > > > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > > > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > > > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > > > > > _______________________________________________ > > > > > Freeswitch-users mailing list > > > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > -- > > > > Anthony Minessale II > > > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > > ClueCon http://www.cluecon.com/ > > > > > > > > AIM: anthm > > > > MSN:anthony_minessale at hotmail.com > > > >< > > > > > MSN%3Aanthony_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 > > > >< > > > > > sip%3A888 at conference.freeswitch.org > > > > > > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > > > > > > > > > > > pstn:213-799-1400 > > > > > > -- > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > -- > Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: > http://www.gmx.net/de/go/multimessenger01 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/3cd27146/attachment-0001.html From vince.freeswitch at hightek.org Mon Mar 30 22:12:37 2009 From: vince.freeswitch at hightek.org (Vincent) Date: Tue, 31 Mar 2009 00:12:37 -0500 Subject: [Freeswitch-users] Errors compiling on dragonfly bsd Message-ID: <20090331051237.GA23340@quark.hightek.org> Hi. I'm trying to compile FS on dragonfly BSD 1.10.1-RELEASE. I am compiling the release source from http://files.freeswitch.org/freeswitch-1.0.3.tar.gz I had to write a uname wrapper script to fake being FreeBSD to get configure to complete successfully. I also added "-D__FreeBSD__" to CPPFLAGS to get past an initial compilation error. Also, I initially got the error, "/usr/pkg/bin/bash: line 8: svnversion: command not found" just prior to the line that appears to generate "src/include/switch_swigable_cpp.h". It seems strange that I would need subversion to compile a release source but I installed it anyway in case that error caused other problems. However, now I get the following errors (which I also got prior to installing subversion). Compiling src/switch_apr.c ... src/switch_apr.c: In function `switch_thread_self': src/switch_apr.c:74: warning: implicit declaration of function `apr_os_thread_current' src/switch_apr.c:74: warning: return makes pointer from integer without a cast src/switch_apr.c: In function `switch_thread_rwlock_create': src/switch_apr.c:221: warning: implicit declaration of function `apr_thread_rwlock_create' src/switch_apr.c: In function `switch_thread_rwlock_destroy': src/switch_apr.c:226: warning: implicit declaration of function `apr_thread_rwlock_destroy' ... Pluss a bunch more apr_thread... related warnings Then gmake[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 Making all in src Making all in mod making all mod_amr gmake[5]: *** No rule to make target `/u1/ports/freeswitch-1.0.3/work/freeswitch-1.0.3/libfreeswitch.la', needed by `mod_amr.so'. Stop. gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_amr-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 *** make failed for freeswitch-1.0.3 *** *** Error code 5 Stop in /u1/ports/freeswitch-1.0.3. It does not seem to check the error code either and still gives a "Build Complete" message. Any help would be appreciated. From raffaele.p.guidi at gmail.com Tue Mar 31 00:56:30 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 31 Mar 2009 09:56:30 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <20090331072858.GA15463@jdc.jasonjgw.net> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <20090331072858.GA15463@jdc.jasonjgw.net> Message-ID: I am a Yate user and I can tell their mailing list suffer the same problem. My solution? I often ask for help but, as a personal policy, I always write an article or add to an existing one on the project wiki explaining and documenting what people explained to me. This creates a triple value: 1. I have stuff explained 2. other people can find this explanation just googling around 3. I don't have to mantain a separate documentation for myself, I just keep referring to (google and) the project wiki (when I don't exactly remember things I sometimes end reading the wiki and saying "oh, I wrote this!") I suggest, in the end, a kind of "ok I'll give you help but you write this stuff in a piece of documentation" policy. I got the name, too: "The Wiki Tax" ;) Regards, Raffaele On Tue, Mar 31, 2009 at 09:28, Jason White wrote: > Brian West wrote: > > This isn't a buffet where you pull up and demand things be one way or the > > other... this is a community where you start helping. I would love to > > see more helping and less demanding! > > So would I. > > I regularly scan the mailing list looking for questions to answer, but many > of > them relate to scenarios of which I have had no experience, or features > that I > haven't had any reason to use. > > I am worried that the questions are answered by the same people much of the > time, which in the long run will be bad for the project as the community > grows, i.e., more technical support work for the same people (who are > mostly > the core developers as well) is not a sustainable proposition. > > How do other large projects handle this? > > Does anyone have any positive suggestions to offer that would encourage > more > contributions? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/55c588ec/attachment.html From mattdfong at gmail.com Tue Mar 31 08:15:06 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 31 Mar 2009 22:15:06 +0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> Message-ID: <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> Got a few more questions about running LUA scripts, please forgive me, I'm an absolute newbie with LUA. If I want to subscribe to a custom event, and I use con = freeswitch.EventConsumer("CUSTOM my::event"); I get an error. Is this because I must subscribe to the CUSTOM (only) event, and then filter out the events using the Event-Subclass myself? Or am I missing something in the syntax of the subscribe? Also, if I do not have a freeswitch.Session, what is the best way to have my LUA script sleep? I want a functionality, where a statement inside my LUA script gets iterated every 30 seconds. My program does not use a session, so I cannot use session:execute("sleep","1000"), as suggested in the wiki. I tried api::sleep(30000) and a few other combinations with execute but no luck :(. Thanks. --matt On Sat, Mar 28, 2009 at 12:43 AM, Michael Collins wrote: > > con = freeswitch.EventConsumer("all"); > > > > now you have a consumer obj > > > > every time you call con:pop() with no arg you will either get an event or > > nil when there are no events to consume. > > every time you call con:pop(1) the consumer object will block until there > is > > an event. > > > > So you use the first way in conjunction with some other lock to do async > or > > the 2nd way you do a dedicated blocking loop. > > FYI, I added this information to the wiki page for > freeswitch.EventConsumer. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/6dd67954/attachment.html From mike at jerris.com Tue Mar 31 08:32:23 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Mar 2009 11:32:23 -0400 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> Message-ID: as replied earlier, if your doing nothing but consuming events, you can just block instead of sleep: con:pop(1) there is also a msleep function that you can call the same way you do console_log, it takes milli seconds as its arg. Note this should NOT be used when you have a script running as a session, only when you are running an api script. Mike On Mar 31, 2009, at 11:15 AM, Matthew Fong wrote: > Got a few more questions about running LUA scripts, please forgive > me, I'm an absolute newbie with LUA. > > If I want to subscribe to a custom event, and I use > > con = freeswitch.EventConsumer("CUSTOM my::event"); > > I get an error. Is this because I must subscribe to the CUSTOM > (only) event, and then filter out the events using the Event- > Subclass myself? Or am I missing something in the syntax of the > subscribe? > > Also, if I do not have a freeswitch.Session, what is the best way to > have my LUA script sleep? I want a functionality, where a statement > inside my LUA script gets iterated every 30 seconds. My program does > not use a session, so I cannot use session:execute("sleep","1000"), > as suggested in the wiki. I tried api::sleep(30000) and a few other > combinations with execute but no luck :(. > > Thanks. > --matt > > On Sat, Mar 28, 2009 at 12:43 AM, Michael Collins > wrote: > > con = freeswitch.EventConsumer("all"); > > > > now you have a consumer obj > > > > every time you call con:pop() with no arg you will either get an > event or > > nil when there are no events to consume. > > every time you call con:pop(1) the consumer object will block > until there is > > an event. > > > > So you use the first way in conjunction with some other lock to do > async or > > the 2nd way you do a dedicated blocking loop. > > FYI, I added this information to the wiki page for > freeswitch.EventConsumer. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/726287a1/attachment.html From mattdfong at gmail.com Tue Mar 31 08:50:28 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 31 Mar 2009 22:50:28 +0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> Message-ID: <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> I know before I asked about blocking for an event, and maybe I should have created a new topic.. but now I want to actually sleep (rather than block) for a set time frame...this app will not be consuming events. can I get an example of how to use msleep in a lua script? This lua script will be running in the background, and not part of a session or event consumer. Thanks. --matt 2009/3/31 Michael Jerris > as replied earlier, if your doing nothing but consuming events, you can > just block instead of sleep: > con:pop(1) > > there is also a msleep function that you can call the same way you do > console_log, it takes milli seconds as its arg. Note this should NOT be > used when you have a script running as a session, only when you are running > an api script. > > Mike > > On Mar 31, 2009, at 11:15 AM, Matthew Fong wrote: > > Got a few more questions about running LUA scripts, please forgive me, I'm > an absolute newbie with LUA. > If I want to subscribe to a custom event, and I use > > con = freeswitch.EventConsumer("CUSTOM my::event"); > > I get an error. Is this because I must subscribe to the CUSTOM (only) > event, and then filter out the events using the Event-Subclass myself? Or > am I missing something in the syntax of the subscribe? > > Also, if I do not have a freeswitch.Session, what is the best way to have > my LUA script sleep? I want a functionality, where a statement inside my LUA > script gets iterated every 30 seconds. My program does not use a session, so > I cannot use session:execute("sleep","1000"), as suggested in the wiki. I > tried api::sleep(30000) and a few other combinations with execute but no > luck :(. > > Thanks. > --matt > > On Sat, Mar 28, 2009 at 12:43 AM, Michael Collins wrote: > >> > con = freeswitch.EventConsumer("all"); >> > >> > now you have a consumer obj >> > >> > every time you call con:pop() with no arg you will either get an event >> or >> > nil when there are no events to consume. >> > every time you call con:pop(1) the consumer object will block until >> there is >> > an event. >> > >> > So you use the first way in conjunction with some other lock to do async >> or >> > the 2nd way you do a dedicated blocking loop. >> >> FYI, I added this information to the wiki page for >> freeswitch.EventConsumer. >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/beaa744d/attachment-0001.html From mike at jerris.com Tue Mar 31 08:59:08 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Mar 2009 11:59:08 -0400 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> Message-ID: <33A62996-1A30-45F1-95C3-A421867CDFF2@jerris.com> http://wiki.freeswitch.org/wiki/Lua#freeswitch.consoleLog On Mar 31, 2009, at 11:50 AM, Matthew Fong wrote: > I know before I asked about blocking for an event, and maybe I > should have created a new topic.. > > but now I want to actually sleep (rather than block) for a set time > frame...this app will not be consuming events. > > can I get an example of how to use msleep in a lua script? This lua > script will be running in the background, and not part of a session > or event consumer. Thanks. > > --matt > > 2009/3/31 Michael Jerris > as replied earlier, if your doing nothing but consuming events, you > can just block instead of sleep: > > con:pop(1) > > there is also a msleep function that you can call the same way you > do console_log, it takes milli seconds as its arg. Note this should > NOT be used when you have a script running as a session, only when > you are running an api script. > > Mike > > On Mar 31, 2009, at 11:15 AM, Matthew Fong wrote: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/05d2754b/attachment.html From brian at freeswitch.org Tue Mar 31 09:02:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 11:02:26 -0500 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> Message-ID: <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> Lua has no sleep or pause ... if you read thru the lua wiki they show you various ways to accomplish that. On Mar 31, 2009, at 10:50 AM, Matthew Fong wrote: > I know before I asked about blocking for an event, and maybe I > should have created a new topic.. > > but now I want to actually sleep (rather than block) for a set time > frame...this app will not be consuming events. > > can I get an example of how to use msleep in a lua script? This lua > script will be running in the background, and not part of a session > or event consumer. Thanks. > > --ma Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/1de2cd0f/attachment.html From mattdfong at gmail.com Tue Mar 31 09:11:29 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 31 Mar 2009 23:11:29 +0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> Message-ID: <4256bf830903310911o143e2ef5o651415c087c7b3c0@mail.gmail.com> Thanks, the freeswitch.msleep(5000) works! Any comment about the first Q... con = freeswitch.EventConsumer("CUSTOM my::event"); I get an error. Is this because I must subscribe to the CUSTOM (only) event, and then filter out the events using the Event-Subclass myself? Or am I missing something in the syntax of the subscribe? Thanks Michael for your help... --matt 2009/3/31 Brian West > Lua has no sleep or pause ... if you read thru the lua wiki they show you > various ways to accomplish that. > On Mar 31, 2009, at 10:50 AM, Matthew Fong wrote: > > I know before I asked about blocking for an event, and maybe I should have > created a new topic.. > but now I want to actually sleep (rather than block) for a set time > frame...this app will not be consuming events. > > can I get an example of how to use msleep in a lua script? This lua script > will be running in the background, and not part of a session or event > consumer. Thanks. > > --ma > > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/f779b7aa/attachment.html From mike at jerris.com Tue Mar 31 09:18:42 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Mar 2009 12:18:42 -0400 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903310911o143e2ef5o651415c087c7b3c0@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> <4256bf830903310911o143e2ef5o651415c087c7b3c0@mail.gmail.com> Message-ID: http://www.freeswitch.org/docs/class_event_consumer.html#a0 It takes 2 args, not one to specify the subclass Mike On Mar 31, 2009, at 12:11 PM, Matthew Fong wrote: > Thanks, the freeswitch.msleep(5000) works! > > Any comment about the first Q... > > con = freeswitch.EventConsumer("CUSTOM my::event"); > > I get an error. Is this because I must subscribe to the CUSTOM > (only) event, and then filter out the events using the Event- > Subclass myself? Or am I missing something in the syntax of the > subscribe? > > > Thanks Michael for your help... > From msc at freeswitch.org Tue Mar 31 09:24:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Mar 2009 09:24:58 -0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> Message-ID: <87f2f3b90903310924v61f559cfx68809ada6ad8f5ed@mail.gmail.com> FYI, I've documented the msleep method here: http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep I will work on better and more organized API documentation. If anyone out there has time/energy/knowledge of the scripting APIs and is willing to help out please email me off list. -MC 2009/3/31 Brian West > Lua has no sleep or pause ... if you read thru the lua wiki they show you > various ways to accomplish that. > On Mar 31, 2009, at 10:50 AM, Matthew Fong wrote: > > I know before I asked about blocking for an event, and maybe I should have > created a new topic.. > but now I want to actually sleep (rather than block) for a set time > frame...this app will not be consuming events. > > can I get an example of how to use msleep in a lua script? This lua script > will be running in the background, and not part of a session or event > consumer. Thanks. > > --ma > > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/c1bb3cb5/attachment.html From msc at freeswitch.org Tue Mar 31 09:26:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Mar 2009 09:26:35 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <20090331072858.GA15463@jdc.jasonjgw.net> Message-ID: <87f2f3b90903310926x21b28c7l116fce1a69e8edff@mail.gmail.com> 2009/3/31 Raffaele P. Guidi > I am a Yate user and I can tell their mailing list suffer the same problem. > My solution? I often ask for help but, as a personal policy, I always write > an article or add to an existing one on the project wiki explaining and > documenting what people explained to me. This creates a triple value: > > 1. I have stuff explained > 2. other people can find this explanation just googling around > 3. I don't have to mantain a separate documentation for myself, I > just keep referring to (google and) the project wiki (when I don't exactly > remember things I sometimes end reading the wiki and saying "oh, I wrote > this!") > > I suggest, in the end, a kind of "ok I'll give you help but you write this > stuff in a piece of documentation" policy. I got the name, too: "The Wiki > Tax" ;) > I like it... "The Wiki Tax" -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/f651715e/attachment-0001.html From gkuri at ieee.org Tue Mar 31 09:59:15 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 31 Mar 2009 09:59:15 -0700 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49C45508.402@coppice.org> References: <49C40A83.1050003@ieee.org> <49C45508.402@coppice.org> Message-ID: <49D24BE3.2020101@ieee.org> > This part is interesting, and the subject of a discussion we had > recently. A number of systems try that second re-invite after a 488, but > the SIP specs say the call is pretty much dead after the 488 message is > exchanged. Are they just hoping that maybe the other end will be > non-compliant enough to keep the call alive, and recover its media mode, > or haven't they read the specs? I think they're hoping the other end is willing to recover it's media mode rather than fail the call. I had no idea the call is technically dead after the 488. Honestly, it would be nice if the call would still be recoverable after that 488 on the T.38 ReINVITE, in order to try and negotiate PCMU to try and keep the FAX going, but if that's not how it's supposed to work, I'd rather follow the spec. So for now I've disabled T.38 completely on both the SPA side and I had the carrier disable it on my trunk, so they won't try a T.38 reinvite. Instead they're trying a PCMU ReINVITE and the problem I'm seeing is that if the carrier reINVITEs PCMU after the call initially started out as G729, FS fails the call, because it seems to be trying to transcode between G729 and PCMU, rather than pass the PCMU reINVITE through to the other leg. 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1550 sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/cedarwireless.net/1909XXXXXXX at 65.98.2 36.38 PCMU/8000 20 ms 160 samples 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() Audio params are unchanged for sofia/cedarwireless.net/ 1909XXXXXXX at 65.98.236.38. 2009-03-31 00:30:50 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing Reinvite 2009-03-31 00:30:50 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/cedarwireless.net/1909XXXXXXX at 65.98.236.38 en tering state [completed] 2009-03-31 00:30:50 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/cedarwireless.net/1909XXXXXXX at 65.98.23 6.38 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-03-31 00:30:50 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-31 00:30:50 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! I have inherit_codec=true, set in the dialplan and disable-transcoding=true set in the sofia profile, which is what I thought would do the trick, but it doesn't seem to be doing anything, FS is still trying to transcode between G729 and PCMU. Is there something I'm missing to get the PCMU ReINVITE from one of the legs to passthrough to the other leg? Does this work only in proxy_media or bypass_media modes? I am testing this with the latest rev of trunk as well. Thanks, Gabe From anthony.minessale at gmail.com Tue Mar 31 12:19:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 14:19:18 -0500 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49D24BE3.2020101@ieee.org> References: <49C40A83.1050003@ieee.org> <49C45508.402@coppice.org> <49D24BE3.2020101@ieee.org> Message-ID: <191c3a030903311219h6c825256l7a52720cf9eb3508@mail.gmail.com> correct, we *do not* proxy re-invites except when bypass_media or proxy_media is set. On Tue, Mar 31, 2009 at 11:59 AM, Gabriel Kuri wrote: > > This part is interesting, and the subject of a discussion we had > > recently. A number of systems try that second re-invite after a 488, but > > the SIP specs say the call is pretty much dead after the 488 message is > > exchanged. Are they just hoping that maybe the other end will be > > non-compliant enough to keep the call alive, and recover its media mode, > > or haven't they read the specs? > > I think they're hoping the other end is willing to recover it's media > mode rather than fail the call. I had no idea the call is technically > dead after the 488. Honestly, it would be nice if the call would still > be recoverable after that 488 on the T.38 ReINVITE, in order to try and > negotiate PCMU to try and keep the FAX going, but if that's not how it's > supposed to work, I'd rather follow the spec. > > So for now I've disabled T.38 completely on both the SPA side and I had > the carrier disable it on my trunk, so they won't try a T.38 reinvite. > Instead they're trying a PCMU ReINVITE and the problem I'm seeing is > that if the carrier reINVITEs PCMU after the call initially started out > as G729, FS fails the call, because it seems to be trying to transcode > between G729 and PCMU, rather than pass the PCMU reINVITE through to the > other leg. > > 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1550 > sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU > 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1601 > sofia_glue_tech_set_codec() Set Codec > sofia/cedarwireless.net/1909XXXXXXX at 65.98.2 > 36.38 PCMU/8000 20 ms 160 samples > 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() > Audio params are unchanged for sofia/cedarwireless.net/ > 1909XXXXXXX at 65.98.236.38. > 2009-03-31 00:30:50 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing Reinvite > 2009-03-31 00:30:50 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/cedarwireless.net/1909XXXXXXX at 65.98.236.38 en > tering state [completed] > 2009-03-31 00:30:50 [DEBUG] switch_core_io.c:655 > switch_core_session_write_frame() > sofia/cedarwireless.net/1909XXXXXXX at 65.98.23 > 6.38 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > 2009-03-31 00:30:50 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-03-31 00:30:50 [ERR] switch_core_io.c:723 > switch_core_session_write_frame() Codec G.729 decoder error! > > I have inherit_codec=true, set in the dialplan and > disable-transcoding=true set in the sofia profile, which is what I > thought would do the trick, but it doesn't seem to be doing anything, FS > is still trying to transcode between G729 and PCMU. Is there something > I'm missing to get the PCMU ReINVITE from one of the legs to passthrough > to the other leg? Does this work only in proxy_media or bypass_media modes? > > I am testing this with the latest rev of trunk as well. > > Thanks, > > Gabe > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/47736722/attachment.html From gmaruzz at celliax.org Tue Mar 31 13:09:05 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 31 Mar 2009 22:09:05 +0200 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <22800505.post@talk.nabble.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> Message-ID: <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> Ciao Bipin, there is both video and audio. Use vlc (http://www.videolan.org/vlc/) or mplayer (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). 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 2009/3/31 xbipin : > > hi, > > is it just audio or is it that im having broken codecs so cant view any > video? > > Regards, > Bipin > > > > Giovanni Maruzzelli-3 wrote: >> >> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS >> Microsoft Windows >> Download 118MB HD: http://www.celliax.org/final.avi >> >> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p22800505.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Tue Mar 31 13:41:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 15:41:24 -0500 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> Message-ID: <191c3a030903311341r5ff21921l417bd49a095443c@mail.gmail.com> I recommend you reformat this file to be in some format that will play on windows and mac and anything else without having to "get" anything or nobody will watch it =D 2009/3/31 Giovanni Maruzzelli > Ciao Bipin, > there is both video and audio. > > Use vlc (http://www.videolan.org/vlc/) or mplayer > (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). > > 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 > > > > > 2009/3/31 xbipin : > > > > hi, > > > > is it just audio or is it that im having broken codecs so cant view any > > video? > > > > Regards, > > Bipin > > > > > > > > Giovanni Maruzzelli-3 wrote: > >> > >> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS > >> Microsoft Windows > >> Download 118MB HD: http://www.celliax.org/final.avi > >> > >> 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 at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > -- > > View this message in context: > http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p22800505.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/f6bda4ac/attachment.html From nik.middleton at noblesolutions.co.uk Tue Mar 31 13:44:40 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 31 Mar 2009 21:44:40 +0100 Subject: [Freeswitch-users] Injecting audio into live call Message-ID: Hi Guys, I know this sounds an odd question, but I need to inject audio into an outbound call. The reason for this is that for a pre-paid billing app, I need to let the call initiator know they are running out of credit. Is there a facility to do this? Ideally I only want to let the subscriber, I.e. the one paying for the call to hear this. In other words 'you have 2 minutes left for this call' Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/66901bb6/attachment-0001.html From nik.middleton at noblesolutions.co.uk Tue Mar 31 13:45:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 31 Mar 2009 21:45:33 +0100 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> Message-ID: Worked for me, just needed to add the missing codec for media player -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: 31 March 2009 21:09 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS and Skypiax on Windows Video How To Ciao Bipin, there is both video and audio. Use vlc (http://www.videolan.org/vlc/) or mplayer (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). 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 2009/3/31 xbipin : > > hi, > > is it just audio or is it that im having broken codecs so cant view any > video? > > Regards, > Bipin > > > > Giovanni Maruzzelli-3 wrote: >> >> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS >> Microsoft Windows >> Download 118MB HD: http://www.celliax.org/final.avi >> >> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p 22800505.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Tue Mar 31 13:51:40 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 15:51:40 -0500 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> Message-ID: <002033F2-5D55-4127-ADBE-8E7A387F7F62@freeswitch.org> You shouldn't have to go get anything :P If you have to spend time to get something to watch the video it sometimes isn't a good thing... have you tried YouTUBE? http://www.perian.org/ /b On Mar 31, 2009, at 3:45 PM, Nik Middleton wrote: > Worked for me, just needed to add the missing codec for media player > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Giovanni Maruzzelli > Sent: 31 March 2009 21:09 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FS and Skypiax on Windows Video How To > > Ciao Bipin, > there is both video and audio. > > Use vlc (http://www.videolan.org/vlc/) or mplayer > (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). > > 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 > > > > > 2009/3/31 xbipin : >> >> hi, >> >> is it just audio or is it that im having broken codecs so cant view > any >> video? >> >> Regards, >> Bipin >> >> >> >> Giovanni Maruzzelli-3 wrote: >>> >>> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS >>> Microsoft Windows >>> Download 118MB HD: http://www.celliax.org/final.avi >>> >>> 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: > http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p > 22800505.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/36fb2af2/attachment.html From brian at freeswitch.org Tue Mar 31 13:53:00 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 15:53:00 -0500 Subject: [Freeswitch-users] Injecting audio into live call In-Reply-To: References: Message-ID: uuid_displace /b On Mar 31, 2009, at 3:44 PM, Nik Middleton wrote: > Hi Guys, > > I know this sounds an odd question, but I need to inject audio into > an outbound call. The reason for this is that for a pre-paid > billing app, I need to let the call initiator know they are running > out of credit. Is there a facility to do this? Ideally I only want > to let the subscriber, I.e. the one paying for the call to hear > this. In other words ?you have 2 minutes left for this call? > > Regards, > _______ Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/649b3f2d/attachment.html From anthony.minessale at gmail.com Tue Mar 31 14:13:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 16:13:41 -0500 Subject: [Freeswitch-users] Injecting audio into live call In-Reply-To: References: Message-ID: <191c3a030903311413g39786ac5y8ac9106c50dc1235@mail.gmail.com> or, probably better for this situation, uuid_broadcast 2009/3/31 Brian West > uuid_displace > /b > > On Mar 31, 2009, at 3:44 PM, Nik Middleton wrote: > > Hi Guys, > > I know this sounds an odd question, but I need to inject audio into an > outbound call. The reason for this is that for a pre-paid billing app, I > need to let the call initiator know they are running out of credit. Is > there a facility to do this? Ideally I only want to let the subscriber, > I.e. the one paying for the call to hear this. In other words ?you have 2 > minutes left for this call? > > Regards, > _______ > > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/e59e58a3/attachment-0001.html From jforman at wcgltd.com Tue Mar 31 15:04:26 2009 From: jforman at wcgltd.com (Josh Forman) Date: Tue, 31 Mar 2009 18:04:26 -0400 Subject: [Freeswitch-users] Greedy codecs on the outbound side Message-ID: I know that on a sofia profile you can set it to be greedy or generous in codec negotiation on the inbound side. How does freeswitch negotate codecs on the outbound side and what ways are available to affect how this takes place? Is there a way to make the codec preferences of the inbound side the codec preference of the outbound side? If not is there a way to set outbound negotations as greedy vs generous? If we disable transcoding in freeswitch would that in effect make the codec negotated on the inbound side be our most prefered codec on the outbound side's codec negotation? Thank you for your help, Josh From badeguruji at yahoo.com Tue Mar 31 16:27:45 2009 From: badeguruji at yahoo.com (badeguruji) Date: Tue, 31 Mar 2009 16:27:45 -0700 (PDT) Subject: [Freeswitch-users] using freeswitch on ubuntu in home network Message-ID: <902786.43554.qm@web80408.mail.mud.yahoo.com> Hello, Friends, can someone please help me understand what hardware and/or software (like FreeSwitch) do i need, to have a working setup like below: DSL from telco----->switch+NAT------>ubuntu server |__________>voip phone |___________>normal PSTN phone Q. what voip related software will i need to run above setup? Q. what hardware (like phone card etc.) do i need in my PC? Q. will i be able to use my existing phone number? (i do not have voip service right now) Q. will i be able to use my regular phone? or do i need voip phone? I am looking for these featuers: 1. full call records. 2. visual voicemail 3. seperate voicemail box for each family member. 4. call forwarding on each user basis. 5. ability to make more then one outbound call, with having ONLY one home phone number. thank you, Rajeev ________________________________ ~~aapka kalyan ho~~ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/0a45a198/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 19861 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/0a45a198/attachment-0001.jpe From jason at jasonjgw.net Tue Mar 31 16:36:43 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Apr 2009 10:36:43 +1100 Subject: [Freeswitch-users] contributing to the wiki In-Reply-To: <87f2f3b90903310926x21b28c7l116fce1a69e8edff@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <20090331072858.GA15463@jdc.jasonjgw.net> <87f2f3b90903310926x21b28c7l116fce1a69e8edff@mail.gmail.com> Message-ID: <20090331233643.GA7764@jdc.jasonjgw.net> Michael Collins wrote: > I like it... "The Wiki Tax" It's an excellent suggestion. As an aside, would it be possible for the wiki administrator to modify the configuration so that there is a means of subscribing without having to deal with a captcha? For people who can't see the screen, this is a real barrier to participation. From sprice at gmail.com Tue Mar 31 16:47:53 2009 From: sprice at gmail.com (SP) Date: Tue, 31 Mar 2009 18:47:53 -0500 Subject: [Freeswitch-users] Greedy codecs on the outbound side In-Reply-To: References: Message-ID: <7e2ac3270903311647g28dc9b4ck10a8ab7cd47aa89f@mail.gmail.com> http://wiki.freeswitch.org/wiki/Codec_negotiation http://lmgtfy.com/?q=freeswitch+late+negotiation On Tue, Mar 31, 2009 at 17:04, Josh Forman wrote: > I know that on a sofia profile you can set it to be greedy or generous > in codec negotiation on the inbound side. > > How does freeswitch negotate codecs on the outbound side and what ways > are available to affect how this takes place? > Is there a way to make the codec preferences of the inbound side the > codec preference of the outbound side? > If not is there a way to set outbound negotations as greedy vs generous? > If we disable transcoding in freeswitch would that in effect make the > codec negotated on the inbound side be our most prefered codec on the > outbound side's codec negotation? > > Thank you for your help, > > Josh > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From brian at freeswitch.org Tue Mar 31 16:49:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 18:49:06 -0500 Subject: [Freeswitch-users] contributing to the wiki In-Reply-To: <20090331233643.GA7764@jdc.jasonjgw.net> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <20090331072858.GA15463@jdc.jasonjgw.net> <87f2f3b90903310926x21b28c7l116fce1a69e8edff@mail.gmail.com> <20090331233643.GA7764@jdc.jasonjgw.net> Message-ID: <76A67C51-ACDB-4916-9ADE-054D53647AA2@freeswitch.org> On Mar 31, 2009, at 6:36 PM, Jason White wrote: > It's an excellent suggestion. > > As an aside, would it be possible for the wiki administrator to > modify the > configuration so that there is a means of subscribing without having > to deal > with a captcha? Not sure we can do that but I suspect we could flag it in such a way where some users can edit without it... I totally understand the need! ;) > > For people who can't see the screen, this is a real barrier to > participatio Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/dc573141/attachment.html From jason at jasonjgw.net Tue Mar 31 16:49:39 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Apr 2009 10:49:39 +1100 Subject: [Freeswitch-users] using freeswitch on ubuntu in home network In-Reply-To: <902786.43554.qm@web80408.mail.mud.yahoo.com> References: <902786.43554.qm@web80408.mail.mud.yahoo.com> Message-ID: <20090331234939.GA7922@jdc.jasonjgw.net> badeguruji wrote: > Q. what voip related software will i need to run above setup? FreeSWITCH > Q. what hardware (like phone card etc.) do i need in my PC? A digium or similar card, and a SIP phone of course. > Q. will i be able to use my existing phone number? (i do not have voip service right now) > Q. will i be able to use my regular phone? or do i need voip phone? That depends on your SIP provider and the laws in your country regarding transferrability of phone numbers. > > I am looking for these featuers: > 1. full call records. > 2. visual voicemail > 3. seperate voicemail box for each family member. > 4. call forwarding on each user basis. > 5. ability to make more then one outbound call, with having ONLY one home phone number. That shouldn't be difficult. From brian at freeswitch.org Tue Mar 31 16:52:48 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 18:52:48 -0500 Subject: [Freeswitch-users] using freeswitch on ubuntu in home network In-Reply-To: <902786.43554.qm@web80408.mail.mud.yahoo.com> References: <902786.43554.qm@web80408.mail.mud.yahoo.com> Message-ID: First off please no HTML emails.... We have a large community of blind users that might have a hard time reading your messages or replying which they do all the time! ;) Thanks guys. On Mar 31, 2009, at 6:27 PM, badeguruji wrote: > Hello, > > Friends, can someone please help me understand what hardware and/or > software (like FreeSwitch) do i need, to have a working setup like > below: > > DSL from telco----->switch+NAT------>ubuntu server > |__________>voip phone > |___________>normal PSTN phone > > Q. what voip related software will i need to run above setup? You'll need FreeSWITCH, Voip Phone and a PSTN interface. > Q. what hardware (like phone card etc.) do i need in my PC? Various interface cards are outlined on the wiki... http://wiki.freeswitch.org > Q. will i be able to use my existing phone number? (i do not have > voip service right now) If you get a card to interface your server to the PSTN then yes you'll be able to use it. > Q. will i be able to use my regular phone? or do i need voip phone? If it can plugin to an ATA or other such device to let you ring a standard phone. > > I am looking for these featuers: > 1. full call records. Yes > 2. visual voicemail Yes. Via web. > 3. seperate voicemail box for each family member. Yes > 4. call forwarding on each user basis. Yes > 5. ability to make more then one outbound call, with having ONLY one > home phone number. You need a voip provider or one line per concurrent call you wish to make. > > > thank you, > Rajeev > ________________________________ > ~~aapka kalyan ho~~ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com From msc at freeswitch.org Tue Mar 31 17:10:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Mar 2009 17:10:58 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects Message-ID: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a "janitor projects" wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say "janitor" projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of "janitors" and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/a0ee2cb7/attachment.html From mitul at enterux.com Tue Mar 31 18:36:36 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 1 Apr 2009 07:06:36 +0530 Subject: [Freeswitch-users] using freeswitch on ubuntu in home network In-Reply-To: <20090331234939.GA7922@jdc.jasonjgw.net> References: <902786.43554.qm@web80408.mail.mud.yahoo.com> <20090331234939.GA7922@jdc.jasonjgw.net> Message-ID: <7A8719B0-F715-45B4-885F-E3ABBE94701E@enterux.com> Hello, You can't make more than one outbound call if you have only one phone line from bsnl. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 01-Apr-09, at 5:19, Jason White wrote: > badeguruji wrote: > >> Q. what voip related software will i need to run above setup? > > FreeSWITCH >> Q. what hardware (like phone card etc.) do i need in my PC? > > A digium or similar card, and a SIP phone of course. >> Q. will i be able to use my existing phone number? (i do not have >> voip service right now) >> Q. will i be able to use my regular phone? or do i need voip phone? > > That depends on your SIP provider and the laws in your country > regarding > transferrability of phone numbers. >> >> I am looking for these featuers: >> 1. full call records. >> 2. visual voicemail >> 3. seperate voicemail box for each family member. >> 4. call forwarding on each user basis. >> 5. ability to make more then one outbound call, with having ONLY >> one home phone number. > > That shouldn't be difficult. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Tue Mar 31 22:50:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 1 Apr 2009 07:50:36 +0200 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <002033F2-5D55-4127-ADBE-8E7A387F7F62@freeswitch.org> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> <002033F2-5D55-4127-ADBE-8E7A387F7F62@freeswitch.org> Message-ID: <7b197bef0903312250s2afe6346r8da13c848a98e3b9@mail.gmail.com> Oooops, I was not aware you cannot see the video on Windows (I use mplayer and vlc on windows, and never bother to start windows media player :-) ). I agree that the best would be youtube or so. I don't know how to upload video on youtube, and I'll be not in my office for a week. Can one of you kind souls upload the video to youtube? It would be soooo nice! I'll try to do that when I'm back if nobody steps out. gm 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 2009/3/31 Brian West : > You shouldn't have to go get anything :P ?If you have to spend time to get > something to watch the video it sometimes isn't a good thing... have you > tried YouTUBE? > http://www.perian.org/ > /b > On Mar 31, 2009, at 3:45 PM, Nik Middleton wrote: > > Worked for me, just needed to add the missing codec for media player > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Giovanni Maruzzelli > Sent: 31 March 2009 21:09 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FS and Skypiax on Windows Video How To > > Ciao Bipin, > there is both video and audio. > > Use vlc (http://www.videolan.org/vlc/) or mplayer > (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). > > 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 > > > > > 2009/3/31 xbipin : > > hi, > > is it just audio or is it that im having broken codecs so cant view > > any > > video? > > Regards, > > Bipin > > > > Giovanni Maruzzelli-3 wrote: > > Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS > > Microsoft Windows > > Download 118MB HD: http://www.celliax.org/final.avi > > 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > > View this message in context: > > http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p > 22800505.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mszlazak at aol.com Tue Mar 31 23:21:54 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 02:21:54 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> Message-ID: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> First off. I would not call it a "janitors project" since that may offend some. A second problem is your notion that documentation is "not-quite-as-important" a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of "customer service." Good customer service is then a part of "sales and marketing." Much more often than not, It's sales and marketing that is more important to making something a "real product"? than engineering. "Build it and they will come" almost never works. Anyway, I think you need a new name for this project. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org ; freeswitch-dev at lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a "janitor projects" wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say "janitor" projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of "janitors" and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009!? http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/2c867d94/attachment.html From codecomplete at free.fr Sun Mar 1 10:20:54 2009 From: codecomplete at free.fr (Fred) Date: Sun, 01 Mar 2009 19:20:54 +0100 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? Message-ID: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> Hello As an easy way to show a Freeswitch server to prospects, I'm thinking of buying an Asus notebook along with a Sangom USB FXO gateway. www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port If someone's been using those two thingies, I'm curious to know if they happily run Freeswitch, or if I should look for some other hardware? Thank you. From rex.alex345 at yahoo.com Sun Mar 1 11:13:34 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST) Subject: [Freeswitch-users] To do telephony functions from web page Message-ID: <1235934814358-2405620.post@n2.nabble.com> Hi All, I am trying to do the telephonic functions(like dial, hangup, conference and etc.) from a webpage (for a customization) rather than doing it from a soft phone. What would be the optimal way of doing it? Please suggest. Thanks, Rex. -- View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405620.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/6a856e00/attachment-0002.html From krice at freeswitch.org Sun Mar 1 11:17:39 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 01 Mar 2009 13:17:39 -0600 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <1235934814358-2405620.post@n2.nabble.com> Message-ID: Check out ESL for PHP, Perl etc, or you can use mod_xml_rpc to control things.... Both methods work well K From: Rex_Alex Reply-To: Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST) To: Subject: [Freeswitch-users] To do telephony functions from web page Hi All, I am trying to do the telephonic functions(like dial, hangup, conference and etc.) from a webpage (for a customization) rather than doing it from a soft phone. What would be the optimal way of doing it? Please suggest. Thanks, Rex. View this message in context: To do telephony functions from web page Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/da1f2954/attachment-0002.html From rex.alex345 at yahoo.com Sun Mar 1 11:51:18 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Sun, 1 Mar 2009 11:51:18 -0800 (PST) Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: References: <1235934814358-2405620.post@n2.nabble.com> Message-ID: <1235937078290-2405782.post@n2.nabble.com> Hi, Learned how to enable mod_xml_rpc but didn't find any samples. Please post me a sample to send requests(like dial) and receive responses(like uuid) from FreeSWITCH using mod_xml_rpc Please assist. Thanks, Rex. Ken Rice-2 wrote: > > Check out ESL for PHP, Perl etc, or you can use mod_xml_rpc to control > things.... Both methods work well > > K > > > > From: Rex_Alex > Reply-To: > Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST) > To: > Subject: [Freeswitch-users] To do telephony functions from web page > > Hi All, I am trying to do the telephonic functions(like dial, hangup, > conference and etc.) from a webpage (for a customization) rather than > doing > it from a soft phone. What would be the optimal way of doing it? Please > suggest. Thanks, Rex. > > View this message in context: To do telephony functions from web page > > Sent from the freeswitch-users mailing list archive > at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405782.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/eeeb7042/attachment-0002.html From sicfslist at gmail.com Sun Mar 1 11:57:03 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 1 Mar 2009 13:57:03 -0600 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <1235937078290-2405782.post@n2.nabble.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> Message-ID: <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> Rex: The basis for xml_rpc or mod_event is something along the lines of: api $command As an example to originate a call (using xml_rpc or mod_event) you would do: api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml $context What language are you trying to do this in? SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/8edf404e/attachment-0002.html From rex.alex345 at yahoo.com Sun Mar 1 12:10:04 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Sun, 1 Mar 2009 12:10:04 -0800 (PST) Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> Message-ID: <1235938204874-2405845.post@n2.nabble.com> Hi Shelby Ramsey, I would like to do the same in php script. Please post me a sample. Thanks, Rex. Shelby Ramsey wrote: > > Rex: > > The basis for xml_rpc or mod_event is something along the lines of: > > api $command > > As an example to originate a call (using xml_rpc or mod_event) you would > do: > > api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml $context > > What language are you trying to do this in? > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405845.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sicfslist at gmail.com Sun Mar 1 12:19:16 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 1 Mar 2009 14:19:16 -0600 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <1235938204874-2405845.post@n2.nabble.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> <1235938204874-2405845.post@n2.nabble.com> Message-ID: <35b355e90903011219p20ee2e5k9e6d553598393394@mail.gmail.com> Rex, I've never actually used PHP for this type of thing ... but you might want to start by looking here: http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/php/single_command.php?r=12216 or http://wiki.freeswitch.org/wiki/PHP_Event_Socket Good luck. I'm sure some other folks here who use PHP for this type of app will be able to assist more. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/3f2f669d/attachment-0002.html From gmaruzz at celliax.org Sun Mar 1 12:30:25 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 1 Mar 2009 21:30:25 +0100 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? In-Reply-To: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> References: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> Message-ID: <7b197bef0903011230k16afc8d1hd43dd224570787f@mail.gmail.com> I, for one, run often FS on an eeepc900 (one year old). NEver tested max concurrent SIP calls, but for sure is able to run concurrently: - one FS - two SIP calls - two Skypiax calls - two linux Skype client instances - two Skype calls Also, I often use it to generate 6 or 8 concurrent Skype calls. So, taking account of how heavy Skype is, it probably is able to run FS with dozens concurrent SIP calls. Gm On 3/1/09, Fred wrote: > Hello > > As an easy way to show a Freeswitch server to prospects, I'm thinking > of buying an Asus notebook along with a Sangom USB FXO gateway. > > www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port > > If someone's been using those two thingies, I'm curious to know if > they happily run Freeswitch, or if I should look for some other hardware? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device 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 From csorlie at teldio.com Sun Mar 1 10:01:21 2009 From: csorlie at teldio.com (Cameron Sorlie) Date: Sun, 01 Mar 2009 13:01:21 -0500 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD Message-ID: <49AACD71.5080103@teldio.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/dc118387/attachment-0002.html From brian at freeswitch.org Sun Mar 1 13:15:20 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Mar 2009 15:15:20 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <49AACD71.5080103@teldio.com> References: <49AACD71.5080103@teldio.com> Message-ID: <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> If i'm not mistaken those events will have a member-id in them so you can tell who they belong to. /b On Mar 1, 2009, at 12:01 PM, Cameron Sorlie wrote: > Using voice activity detection (VAD) in FreeSWITCH, how might I then > distinguish which side of a call any given TALK or NOTALK event > relates to? I am interested not just that there is activity on the > call, but interested also in which party on the call is speaking (or > not). > > Cam From Prometheus001 at gmx.net Sun Mar 1 13:38:20 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 01 Mar 2009 22:38:20 +0100 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number Message-ID: <49AB004C.6090604@gmx.net> Hello, I have the following problem while providing callback (mod_eventsocket is used): 1) I want to call a certain destination number A with a suppressed caller_id_number (this works fine with some vars in the origination string) 2) The destination number A picks up the phone and enters a target number B by DTMF 3) freeswitch then forwards the call to target number B by DTMF and I want to show the number A. I do this with uuid_setvar. The problem is that it still shows unknown. This is all with SIP. uuid_setvar however worked if I did not set the caller_id_number to unknown. Per default this is then "00000000000" and can then be changed with uuid_setvar to the number of A. But if I set caller_id_number to unknown I can no longer change it to A. Any hint? Best regards Peter From Prometheus001 at gmx.net Sun Mar 1 14:27:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 01 Mar 2009 23:27:25 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: References: <49A92BAE.4090907@gmx.net> Message-ID: <49AB0BCD.8030108@gmx.net> Hello Brian, thanks for the info. I am a step further, but it cannot load the grammar files. I am sending through event_socket: SendMsg call-command: execute execute-app-name: detect_speech execute-app-arg: pocketsphinx yes no However I get the message (also when I am using Pizza demo): 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/1000 at sip2.server.com Command Execute detect_speech(pocketsphinx yes no) 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 pocketsphinx_asr_load_grammar() Can't open language model /usr/local/freeswitch/grammar/model/communicator. 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 switch_ivr_detect_speech() Error loading Grammar 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 pocketsphinx_asr_close() Port Closed. However the grammar files are there: root at sip2:/usr/local/freeswitch/grammar/model/communicator# root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al total 12752 drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances Any hint? Best regards Peter Brian West schrieb: > You can accomplish this .... here is an example using ESL in perl > > http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 > > /b > > On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: > > >> Or back to the basics: Is it possible to use pocketsphinx through >> event >> socket? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mike at jerris.com Sun Mar 1 16:34:30 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 1 Mar 2009 19:34:30 -0500 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <1235938204874-2405845.post@n2.nabble.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> <1235938204874-2405845.post@n2.nabble.com> Message-ID: <2A727DCA-B690-4A30-97EF-4D16223ECE45@jerris.com> There are examples on the wiki for this. Mike On Mar 1, 2009, at 3:10 PM, Rex_Alex wrote: > > Hi Shelby Ramsey, > > I would like to do the same in php script. > > Please post me a sample. > > Thanks, > Rex. > > > Shelby Ramsey wrote: >> >> Rex: >> >> The basis for xml_rpc or mod_event is something along the lines of: >> >> api $command >> >> As an example to originate a call (using xml_rpc or mod_event) you >> would >> do: >> >> api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml >> $context >> >> What language are you trying to do this in? >> >> SDR >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405845.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From red.rain.seven at gmail.com Sun Mar 1 18:20:32 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Sun, 1 Mar 2009 18:20:32 -0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> Message-ID: <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> Does the freeswitch VAD is able to distinguish ring tone from human voice? The scenario is to originate a call to a IVR system(don't connect the other leg here yet) and dial DTMF to get to the designated extension number , once someone picks up and say hello ( detected by VAD) now release to connect the other leg of the call. The point is to hold the first leg till a real person picks up. If it can't be done by VAD, how should I approach this function that I want to achieve. Thanks On Sun, Mar 1, 2009 at 1:15 PM, Brian West wrote: > If i'm not mistaken those events will have a member-id in them so you > can tell who they belong to. > > /b > > On Mar 1, 2009, at 12:01 PM, Cameron Sorlie wrote: > > > Using voice activity detection (VAD) in FreeSWITCH, how might I then > > distinguish which side of a call any given TALK or NOTALK event > > relates to? I am interested not just that there is activity on the > > call, but interested also in which party on the call is speaking (or > > not). > > > > Cam > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/d4c7a68c/attachment-0002.html From brian at freeswitch.org Sun Mar 1 18:28:18 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Mar 2009 20:28:18 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> Message-ID: <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> NO. You want something that people THINK exists and works well... Reliable human/voice detection doesn't exist in ANY form. /b On Mar 1, 2009, at 8:20 PM, Henry Huang wrote: > Does the freeswitch VAD is able to distinguish ring tone from human > voice? > The scenario is to originate a call to a IVR system(don't connect > the other leg here yet) and dial DTMF to get to the designated > extension number , once someone picks up and say hello ( detected by > VAD) now release to connect the other leg of the call. The point is > to hold the first leg till a real person picks up. > > If it can't be done by VAD, how should I approach this function that > I want to achieve. > > Thanks From red.rain.seven at gmail.com Sun Mar 1 19:05:19 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Sun, 1 Mar 2009 19:05:19 -0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com> Well, I knew it would be some future fantasy for now.. If not human detection. I guess will try to use Dialplan Tools wait for silence to wait till the ring tone is finished ,then connect the other leg. On Sun, Mar 1, 2009 at 6:28 PM, Brian West wrote: > NO. You want something that people THINK exists and works well... > Reliable human/voice detection doesn't exist in ANY form. > > /b > > On Mar 1, 2009, at 8:20 PM, Henry Huang wrote: > > > Does the freeswitch VAD is able to distinguish ring tone from human > > voice? > > The scenario is to originate a call to a IVR system(don't connect > > the other leg here yet) and dial DTMF to get to the designated > > extension number , once someone picks up and say hello ( detected by > > VAD) now release to connect the other leg of the call. The point is > > to hold the first leg till a real person picks up. > > > > If it can't be done by VAD, how should I approach this function that > > I want to achieve. > > > > Thanks > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/4d9a7260/attachment-0002.html From brian at freeswitch.org Sun Mar 1 19:11:32 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Mar 2009 21:11:32 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com> Message-ID: Usually ringing is done in early media... so the best bet would be to ignore_early_media=true /b On Mar 1, 2009, at 9:05 PM, Henry Huang wrote: > Well, I knew it would be some future fantasy for now.. > If not human detection. I guess will try to use Dialplan Tools wait > for silence to wait till the ring tone is finished ,then connect the > other leg. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/9f2529d5/attachment-0002.html From mrene_lists at avgs.ca Sun Mar 1 21:00:44 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 2 Mar 2009 00:00:44 -0500 Subject: [Freeswitch-users] Qt portaudio interface Message-ID: <3A65F8B6-96B4-45EE-80E9-A30AFEED0BB3@avgs.ca> Hi all, Anyone interested in contributing to a Qt interface in order to make a decent softphone using FS please reply to this thread. (also give your availability so we can have a conference call to decide stuff) Thanks, Math From kawarod at laposte.net Sun Mar 1 23:06:10 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Mar 2009 11:06:10 +0400 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test Message-ID: <49AB8562.4050806@laposte.net> Hi All, I ran some longer tests with FS 1.0.3 acting as an SBC. The test machine has the following specs: - Intel Quad Core Q9550 - 8GB RAM (far too much from what I saw) After 3 days running SIPP with 750 simultaneous calls (1500 channels) at 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. In the CLI, using status command I got this: freeswitch at internal> status UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds, 607 microseconds 15817560 session(s) since startup 22 session(s) 0/500 But when I use "show channels" or "show calls", I see nothing. So I'm wondering where are these 22 sessions ? FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs. Successful call --> 5271434 Failed call ---> 1554 (less than 0.03%) regards, rod. complete SIPP summary: ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 10.10.10.254:5060(UDP) 0 new calls during 0.856 s period 7 ms scheduler resolution 0 calls (limit 750) Peak was 750 calls, after 15 s 0 Running, 34 Paused, 0 Woken up 15544 out-of-call msg (discarded) 1 open sockets 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate (kB/s) 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate (kB/s) Messages Retrans Timeout Unexpected-Msg INVITE ----------> 5273022 0 0 100 <---------- 5273022 0 1554 180 <---------- 0 0 0 183 <---------- 0 0 0 200 <---------- E-RTD1 5271434 0 0 ACK ----------> 5271434 0 Pause [ 35.0s] 5271434 0 BYE ----------> 5271434 0 0 200 <---------- 5271434 0 0 ------------------------------ Test Terminated -------------------------------- ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2009-02-27 09:11:31 Last Reset Time | 2009-03-02 07:49:10 Current Time | 2009-03-02 07:49:11 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:857 | 70:37:39:429 Call Rate | 0.000 cps | 20.739 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 5273022 Total Call created | | 5273022 Current Call | 34 | -------------------------+---------------------------+-------------------------- Successful call | 0 | 5271434 Failed call | 0 | 1554 -------------------------+---------------------------+-------------------------- Response Time 1 | 00:00:00:000 | 00:00:00:240 Call Length | 38:32:13:386 | 00:00:36:131 ------------------------------ Test Terminated -------------------------------- From red.rain.seven at gmail.com Sun Mar 1 23:18:02 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Sun, 1 Mar 2009 23:18:02 -0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com> Message-ID: <59ad9ca10903012318k53852016ied8d982a467577c3@mail.gmail.com> ignore_early_media=true is not going to do the trick since once the IVR picks up the call on leg A, the ring tone is stopped and the IVR is going to play pre-recorded voice menu. And the freeswtich is going to send DTMF to reach a certain extension number say 101. Then the ring tone is going to start again while the IVR is going to dial the 101 extension(or even play moh while dialing). After extension 101 picks up, this is when I want the "originate" to connect call leg B on some other number. On Sun, Mar 1, 2009 at 7:11 PM, Brian West wrote: > Usually ringing is done in early media... so the best bet would be to > ignore_early_media=true > /b > > On Mar 1, 2009, at 9:05 PM, Henry Huang wrote: > > Well, I knew it would be some future fantasy for now.. > If not human detection. I guess will try to use Dialplan Tools wait for > silence to wait till the ring tone is finished ,then connect the other leg. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/ebab2445/attachment-0002.html From saigop at gmail.com Mon Mar 2 01:25:23 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Mon, 2 Mar 2009 14:55:23 +0530 Subject: [Freeswitch-users] To do telephony functions from web page In-Reply-To: <2A727DCA-B690-4A30-97EF-4D16223ECE45@jerris.com> References: <1235934814358-2405620.post@n2.nabble.com> <1235937078290-2405782.post@n2.nabble.com> <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com> <1235938204874-2405845.post@n2.nabble.com> <2A727DCA-B690-4A30-97EF-4D16223ECE45@jerris.com> Message-ID: <2ea4d47e0903020125m4be4a5ffl4c33e6f3325a2919@mail.gmail.com> Hi Rex, Please find the attached file for the PHP script. This script has been executed in FS 1.0.2. put these two scripts in htdocs directory. access the http://localhost/sample2.php so that two text box will appear. you can able to give the extension number and mobile number to dial. Try this :) On Mon, Mar 2, 2009 at 6:04 AM, Michael Jerris wrote: > There are examples on the wiki for this. > > Mike > > On Mar 1, 2009, at 3:10 PM, Rex_Alex wrote: > > > > > Hi Shelby Ramsey, > > > > I would like to do the same in php script. > > > > Please post me a sample. > > > > Thanks, > > Rex. > > > > > > Shelby Ramsey wrote: > >> > >> Rex: > >> > >> The basis for xml_rpc or mod_event is something along the lines of: > >> > >> api $command > >> > >> As an example to originate a call (using xml_rpc or mod_event) you > >> would > >> do: > >> > >> api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml > >> $context > >> > >> What language are you trying to do this in? > >> > >> SDR > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > -- > > View this message in context: > http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405845.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/ae5592d0/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sample2.php Type: application/octet-stream Size: 405 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/ae5592d0/attachment-0004.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: testsample.php Type: application/octet-stream Size: 1434 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/ae5592d0/attachment-0005.obj From gopal2krishnan at gmail.com Mon Mar 2 01:29:13 2009 From: gopal2krishnan at gmail.com (Gopal krishnan) Date: Mon, 2 Mar 2009 14:59:13 +0530 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <1235740392995-2395557.post@n2.nabble.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <49A6E1DB.3070806@freeswitch.org> <1235740392995-2395557.post@n2.nabble.com> Message-ID: <2ea4d47e0903020129y676bbc35n8727f8906237edce@mail.gmail.com> Hi, Actually what is the difference between ESL in FS 1.0.3 and event socket in FS 1.0.2. Is the FS 1.0.3 ESL superior? On Fri, Feb 27, 2009 at 6:43 PM, Rex_Alex wrote: > Hi All, I did what you have all suggested. Now its working perfectly. > Thanks a lot for all your assistance. Rex. > > Raymond Chandler wrote: > and it will probably be a good idea to do make phpmod-install so that the > .so and .php files gets into the correct place to be included -Ray Mathieu > Rene wrote: > > You need your distro's php dev pakage. > On 26-Feb-09, at > 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh && > ./configure && make >> install and did Mathieu's suggestion but getting > error as below, >> [root at server esl]# make phpmod make MYLIB="../libesl.a" > >> SOLINK="-shared -Xlinker -x" >> > CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >> -g > -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >> -Werror > -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> > -Wmissing-prototypes" >> > CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >> -DHAVE_EDITLINE > -g -ggdb -I../../libs/libedit/src/ -fPIC" >> CXX_CFLAGS="" -C php make[1]: > php-config: Command not found make[1]: >> Entering directory > `/root/freeswitch-1.0.3/libs/esl/php' g++ >> > -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >> -ggdb > -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >> > esl_wrap.cpp:717:18: error: zend.h: No such file or directory >> > esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >> > esl_wrap.cpp:719:17: error: php.h: No such file or directory >> > esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >> > esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >> > directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >> > scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >> > ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >> > ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >> error: > expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >> variable or > field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error: > ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was > not declared in this scope >> esl_wrap.cpp:793: error: expected > primary-expression before ?void? >> esl_wrap.cpp:793: error: expected > primary-expression before ?*? token >> esl_wrap.cpp:793: error: ?type? was > not declared in this scope >> esl_wrap.cpp:793: error: expected > primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer > expression list treated as >> compound expression esl_wrap.cpp:793: error: > expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 > make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: > *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i > wrong? Thanks, Rex >> >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34 > AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it.. > Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh && > ./configure && >> make install And then do Mathieu's suggestion? -MC >> > _______________________________________________ Freeswitch-users >> mailing > list Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>> > http://www.freeswitch.org >> >> >> > ------------------------------------------------------------------------ >> > View this message in context: Re: ESL Wrapper >> >> Sent from the > freeswitch-users mailing list archive >> at Nabble.com. >> > _______________________________________________ >> Freeswitch-users mailing > list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > View this message in context: Re: ESL Wrapper > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/fb354cfa/attachment-0002.html From gopal2krishnan at gmail.com Mon Mar 2 01:42:19 2009 From: gopal2krishnan at gmail.com (Gopal krishnan) Date: Mon, 2 Mar 2009 15:12:19 +0530 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? In-Reply-To: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> References: <7.0.1.0.2.20090301191907.027bd870@fredshack.com> Message-ID: <2ea4d47e0903020142n62479f72o996243ea6b5bee64@mail.gmail.com> Hi Fred, Yes you can use Sangoma USB FXO with your laptop. You need to install openzap for this. But for testing you can use this driver. Still there is some issue with Openzap with FS as for as I used. while installing Sangoma USB FXO device you need to use beta drivers. On Sun, Mar 1, 2009 at 11:50 PM, Fred wrote: > Hello > > As an easy way to show a Freeswitch server to prospects, I'm thinking > of buying an Asus notebook along with a Sangom USB FXO gateway. > > www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port > > If someone's been using those two thingies, I'm curious to know if > they happily run Freeswitch, or if I should look for some other hardware? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/d36552a8/attachment-0002.html From codecomplete at free.fr Mon Mar 2 03:47:19 2009 From: codecomplete at free.fr (Fred) Date: Mon, 02 Mar 2009 12:47:19 +0100 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? In-Reply-To: References: Message-ID: <7.0.1.0.2.20090302124612.028657a8@free.fr> Thanks guys for the feedback. So, the OpenZap driver isn't ready for production yet? From sridhart at alcatel-lucent.com Mon Mar 2 03:52:10 2009 From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar)) Date: Mon, 2 Mar 2009 17:22:10 +0530 Subject: [Freeswitch-users] Running freeswitch on powerpc In-Reply-To: References: Message-ID: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> Hi all, I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code Regards Sridhar > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On > Behalf Of freeswitch-users-request at lists.freeswitch.org > Sent: Monday, February 02, 2009 9:12 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 32, Issue 17 > > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more > specific than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: Call Variable not available when call hangup (shehzad p) > 2. Re: How do I set my FS internal ip address to a "static" > value. (clif at eugeneweb.com) > 3. Re: Call Variable not available when call hangup > (Anthony Minessale) > 4. Re: How do I set my FS internal ip address to a "static" > value. (Brian West) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST) > From: shehzad p > Subject: Re: [Freeswitch-users] Call Variable not available when call > hangup > To: freeswitch-users at lists.freeswitch.org > Message-ID: <21791503.post at talk.nabble.com> > Content-Type: text/plain; charset=us-ascii > > > > one question is that when javascript is being called from > dial plan, I get the session object already available, It is > for A leg of channel, So when javascript is called after > Bridge how can I get the session object for B leg also? > > > Anthony Minessale-2 wrote: > > > > the leg you are running the script on is not hungup, the > other leg of the > > call is. > > > > If it was hungup you would not be executing the script. > > > > Asterisk and the h ext and the whole dead-agi thing are all > poor design > > showing it's teeth. > > We do not support anything like it. > > > > > > You can however try this: (see the link below) > > > > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks > -giving-me-headaches-p21614840.html > > > > > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: > > > >> > >> Is there any settings that when call hangup control can be > transferred to > >> another context and these CDR values can be accessible > there? (just like > >> in > >> Asterisk, h extension) > >> > >> shehzad p wrote: > >> > > >> > Hi all, > >> > > >> > I need to process some CDR variables in Dialplan, like > call duration, > >> > Answered time etc. > >> > but when I place info application after bridge, it is > not listing them > >> > properly as below: > >> > =========================================== > >> > Caller-Channel-Created-Time: [1233573341672157] > >> > Caller-Channel-Answered-Time: [1233573342712939] > >> > Caller-Channel-Hangup-Time: [0] > >> > ========================================== > >> > Here Hangup time is 0, So how can I find actual values? > >> > > >> > --I know that we can use xml_cdr or cdr_csv, but my > current need is to > >> get > >> > those values from dialplan itself so that can be passed to some > >> script... > >> > > >> > > >> > thanks, > >> > msp > >> > > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21789152.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com ny.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 f%2B888 at conference.freeswitch.org> > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21791503.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > ------------------------------ > > Message: 2 > Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST) > From: clif at eugeneweb.com > Subject: Re: [Freeswitch-users] How do I set my FS internal ip address > to a "static" value. > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > Hi Gang, > > I've been struggleing with this also. Actually I can get it > to bind to my > address, the problem is it randomly drops my calls. :-( > > I have a FS running on a box with a static IP and I can start > a call between > two extensions and it will go for hours. Then I add anther > interface say eth0:0 > with a new static IP and reconfigure my phones and FS to use > that, and the > calls drop after about 15-20 mins. Though it's pretty random. > > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and > freeswitch-1.0.1. > Here is my /etc/network/interfaces: > > # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) > > # The loopback interface > auto lo > iface lo inet loopback > > # The first network card - this entry was created during the Debian > installation > auto eth0 eth0:0 > iface eth0 inet dhcp > iface eth0:0 inet static > address 192.168.0.249 > netmask 255.255.255.0 > gateway 192.168.0.254 > > The only change I made to the FS config is in Vars.xml. I > added this line close > to the top: > > > > Here is the console log of the call being dropped: > > freeswitch at archive> sofia status > API CALL [sofia(status)] output: > Name Type > Data > State > ============================================================== > =================================== > external profile > sip:mod_sofia at 67.171.158.226:5080 > RUNNING (0) > internal profile > sip:mod_sofia at 192.168.0.249:5060 > RUNNING (2) > nat profile > sip:mod_sofia at 67.171.158.226:5070 > RUNNING (0) > default alias > internal > ALIASED > outbound alias > external > ALIASED > 192.168.0.249 alias > internal > ALIASED > ============================================================== > =================================== > 3 profiles 3 aliases > > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 > sofia_glue_restart_all_profiles() Reload XML [Success] > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() > ENUM Reloaded > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 > sofia_read_frame() Hangup > sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp > ;fs_nat=yes > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 > switch_ivr_multi_threaded_bridge() Hangup > sofia/internal/1001 at 192.168.0.249 > [CS_EXECUTE] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 6 > (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud > p;fs_nat=yes) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel > sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp > ;fs_nat=yes > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 5 > (sofia/internal/1001 at 192.168.0.249) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel > sofia/internal/1001 at 192.168.0.249 > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [192.168.0.249] for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() > Adding Alias [default] > for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() > Started Profile > internal [sofia_reg_internal] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [outbound] for profile [external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() > Started Profile > external [sofia_reg_external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:645 > sofia_profile_thread_run() waiting for > worker thread > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() > Started Profile nat > [sofia_reg_nat] > sofia status > API CALL [sofia(status)] output: > Name Type > Data > State > ============================================================== > =================================== > external profile > sip:mod_sofia at 67.171.158.226:5080 > RUNNING (0) > internal profile > sip:mod_sofia at 192.168.0.249:5060 > RUNNING (0) > outbound alias > external > ALIASED > 192.168.0.249 alias > internal > ALIASED > nat profile > sip:mod_sofia at 67.171.158.226:5070 > RUNNING (0) > default alias > internal > ALIASED > ============================================================== > =================================== > 3 profiles 3 aliases > > There is an older thread that says one should set > > but in this (later) thread is says only Jingleling usese that > variable. > ie. see: > http://www.mail-archive.com/freeswitch-users at lists.freeswitch. > org/msg00695.html > http://www.mail-archive.com/freeswitch-users at lists.freeswitch. > org/msg07345.html > > So what do you think causes this? What is the correct way? ;-) > > > Thanks, > Clif > > > > > ------------------------------ > > Message: 3 > Date: Mon, 2 Feb 2009 09:41:05 -0600 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Call Variable not available when call > hangup > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > you can't that's why i said it was a horrible approach. > That's also why i posted you the instructions on the only > elegant solution > to your problem. > > > On Mon, Feb 2, 2009 at 9:21 AM, shehzad p wrote: > > > > > > > one question is that when javascript is being called from > dial plan, I get > > the session object already available, It is for A leg of channel, > > So when javascript is called after Bridge how can I get the > session object > > for B leg also? > > > > > > Anthony Minessale-2 wrote: > > > > > > the leg you are running the script on is not hungup, the > other leg of the > > > call is. > > > > > > If it was hungup you would not be executing the script. > > > > > > Asterisk and the h ext and the whole dead-agi thing are > all poor design > > > showing it's teeth. > > > We do not support anything like it. > > > > > > > > > You can however try this: (see the link below) > > > > > > > > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks > -giving-me-headaches-p21614840.html > > > > > > > > > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p > wrote: > > > > > >> > > >> Is there any settings that when call hangup control can > be transferred > > to > > >> another context and these CDR values can be accessible > there? (just like > > >> in > > >> Asterisk, h extension) > > >> > > >> shehzad p wrote: > > >> > > > >> > Hi all, > > >> > > > >> > I need to process some CDR variables in Dialplan, like > call duration, > > >> > Answered time etc. > > >> > but when I place info application after bridge, it is > not listing them > > >> > properly as below: > > >> > =========================================== > > >> > Caller-Channel-Created-Time: [1233573341672157] > > >> > Caller-Channel-Answered-Time: [1233573342712939] > > >> > Caller-Channel-Hangup-Time: [0] > > >> > ========================================== > > >> > Here Hangup time is 0, So how can I find actual values? > > >> > > > >> > --I know that we can use xml_cdr or cdr_csv, but my > current need is to > > >> get > > >> > those values from dialplan itself so that can be passed to some > > >> script... > > >> > > > >> > > > >> > thanks, > > >> > msp > > >> > > > >> > > >> -- > > >> View this message in context: > > >> > > > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21789152.html > > >> Sent from the Freeswitch-users mailing list archive at > Nabble.com. > > >> > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > >> > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > < > > > MSN%3Aanthony_minessale at hotmail.com hotmail.com> > > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com ny.minessale at gmail.com> > > > sale at gmail.com> > > > > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > < > > > sip%3A888 at conference.freeswitch.org eswitch.org> > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org f%2B888 at conference.freeswitch.org> > > > 253Aconf%252B888 at conference.freeswitch.org> > > > > > > pstn:213-799-1400 > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > > http://www.freeswitch.org > > > > > > > > > > -- > > View this message in context: > > > http://www.nabble.com/Call-Variable-not-available-when-call-ha > ngup-tp21788550p21791503.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com ny.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 f%2B888 at conference.freeswitch.org> > pstn:213-799-1400 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachm > ents/20090202/2d430e44/attachment-0001.html > > ------------------------------ > > Message: 4 > Date: Mon, 2 Feb 2009 09:41:39 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] How do I set my FS internal ip address > to a "static" value. > To: freeswitch-users at lists.freeswitch.org > Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > you need to add this setting to sofia.conf.xml > > > > > You'll also need to edit the sofia profiles and input the > exact IP you > wish it to bind to. The params are sip-ip and rtp-ip. > > /b > > On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote: > > > Hi Gang, > > > > I've been struggleing with this also. Actually I can get it > to bind > > to my > > address, the problem is it randomly drops my calls. :-( > > > > I have a FS running on a box with a static IP and I can > start a call > > between > > two extensions and it will go for hours. Then I add anther > interface > > say eth0:0 > > with a new static IP and reconfigure my phones and FS to use that, > > and the > > calls drop after about 15-20 mins. Though it's pretty random. > > > > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and > > freeswitch-1.0.1. > > Here is my /etc/network/interfaces: > > > > # /etc/network/interfaces -- configuration file for > ifup(8), ifdown(8) > > > > # The loopback interface > > auto lo > > iface lo inet loopback > > > > # The first network card - this entry was created during the Debian > > installation > > auto eth0 eth0:0 > > iface eth0 inet dhcp > > iface eth0:0 inet static > > address 192.168.0.249 > > netmask 255.255.255.0 > > gateway 192.168.0.254 > > > > The only change I made to the FS config is in Vars.xml. I > added this > > line close > > to the top: > > > > > > > > Here is the console log of the call being dropped: > > > > freeswitch at archive> sofia status > > API CALL [sofia(status)] output: > > Name Type > Data > > State > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > > ====================================================================== > > external profile > sip:mod_sofia at 67.171.158.226:5080 > > RUNNING (0) > > internal profile > sip:mod_sofia at 192.168.0.249:5060 > > RUNNING (2) > > nat profile > sip:mod_sofia at 67.171.158.226:5070 > > RUNNING (0) > > default alias > internal > > ALIASED > > outbound alias > external > > ALIASED > > 192.168.0.249 alias > internal > > ALIASED > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > > ====================================================================== > > 3 profiles 3 aliases > > > > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 > > sofia_glue_restart_all_profiles() Reload XML [Success] > > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM > > Reloaded > > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 > sofia_read_frame() Hangup > > sofia/internal/ > > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes > > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 > > switch_ivr_multi_threaded_bridge() Hangup > sofia/internal/1001 at 192.168.0.249 > > [CS_EXECUTE] [NORMAL_CLEARING] > > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > > switch_core_session_thread() Session 6 > > (sofia/internal/ > > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) > > Ended > > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > > switch_core_session_thread() Close Channel > > sofia/internal/ > > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes > > [CS_HANGUP] > > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > > switch_core_session_thread() Session 5 > (sofia/internal/1001 at 192.168.0.249 > > ) > > Ended > > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > > switch_core_session_thread() Close Channel > sofia/internal/1001 at 192.168.0.249 > > [CS_HANGUP] > > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() > > waiting for > > worker thread > > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 > sofia_profile_thread_run() > > waiting for > > worker thread > > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() > Adding Alias > > [192.168.0.249] for profile [internal] > > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding > > Alias [default] > > for profile [internal] > > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started > > Profile > > internal [sofia_reg_internal] > > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() > Adding Alias > > [outbound] for profile [external] > > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started > > Profile > > external [sofia_reg_external] > > 2009-02-01 13:23:20 [NOTICE] sofia.c:645 > sofia_profile_thread_run() > > waiting for > > worker thread > > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started > > Profile nat > > [sofia_reg_nat] > > sofia status > > API CALL [sofia(status)] output: > > Name Type > Data > > State > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > > ====================================================================== > > external profile > sip:mod_sofia at 67.171.158.226:5080 > > RUNNING (0) > > internal profile > sip:mod_sofia at 192.168.0.249:5060 > > RUNNING (0) > > outbound alias > external > > ALIASED > > 192.168.0.249 alias > internal > > ALIASED > > nat profile > sip:mod_sofia at 67.171.158.226:5070 > > RUNNING (0) > > default alias > internal > > ALIASED > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > = > > > ====================================================================== > > 3 profiles 3 aliases > > > > There is an older thread that says one should set > > > > but in this (later) thread is says only Jingleling usese that > > variable. > > ie. see: > > > http://www.mail-archive.com/freeswitch-users at lists.freeswitch. > org/msg00695.html > > > http://www.mail-archive.com/freeswitch-users at lists.freeswitch. > org/msg07345.html > > > > So what do you think causes this? What is the correct way? ;-) > > > > > > Thanks, > > Clif > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > > http://www.freeswitch.org > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw > itch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 32, Issue 17 > ************************************************ > From gmaruzz at celliax.org Mon Mar 2 04:11:26 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 2 Mar 2009 13:11:26 +0100 Subject: [Freeswitch-users] Running freeswitch on powerpc In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <7b197bef0903020411q72be83efxace263ab3401c001@mail.gmail.com> On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar) wrote: > I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code Hi Sridhar, I don't think someone has tried that. It will probably be you that let us all know which (if any) changes needs to be done. :-) 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 Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar) wrote: > Hi all, > > I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code > > Regards > Sridhar > > >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On >> Behalf Of freeswitch-users-request at lists.freeswitch.org >> Sent: Monday, February 02, 2009 9:12 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Freeswitch-users Digest, Vol 32, Issue 17 >> >> Send Freeswitch-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more >> specific than "Re: Contents of Freeswitch-users digest..." >> >> >> Today's Topics: >> >> 1. Re: Call Variable not available when call hangup (shehzad p) >> 2. Re: How do I set my FS internal ip address to a "static" >> value. (clif at eugeneweb.com) >> 3. Re: Call Variable not available when call hangup >> (Anthony Minessale) >> 4. Re: How do I set my FS internal ip address to a "static" >> value. (Brian West) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST) >> From: shehzad p >> Subject: Re: [Freeswitch-users] Call Variable not available when call >> hangup >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <21791503.post at talk.nabble.com> >> Content-Type: text/plain; charset=us-ascii >> >> >> >> one question is that when javascript is being called from >> dial plan, I get the session object already available, It is >> for A leg of channel, So when javascript is called after >> Bridge how can I get the session object for B leg also? >> >> >> Anthony Minessale-2 wrote: >> > >> > the leg you are running the script on is not hungup, the >> other leg of the >> > call is. >> > >> > If it was hungup you would not be executing the script. >> > >> > Asterisk and the h ext and the whole dead-agi thing are all >> poor design >> > showing it's teeth. >> > We do not support anything like it. >> > >> > >> > You can however try this: (see the link below) >> > >> > >> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks >> -giving-me-headaches-p21614840.html >> > >> > >> > >> > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: >> > >> >> >> >> Is there any settings that when call hangup control can be >> transferred to >> >> another context and these CDR values can be accessible >> there? (just like >> >> in >> >> Asterisk, h extension) >> >> >> >> shehzad p wrote: >> >> > >> >> > Hi all, >> >> > >> >> > I need to process some CDR variables in Dialplan, like >> call duration, >> >> > Answered time etc. >> >> > but when I place info application after bridge, it is >> not listing them >> >> > properly as below: >> >> > =========================================== >> >> > Caller-Channel-Created-Time: [1233573341672157] >> >> > Caller-Channel-Answered-Time: [1233573342712939] >> >> > Caller-Channel-Hangup-Time: [0] >> >> > ========================================== >> >> > Here Hangup time is 0, So how can I find actual values? >> >> > >> >> > --I know that we can use xml_cdr or cdr_csv, but my >> current need is to >> >> get >> >> > those values from dialplan itself so that can be passed to some >> >> script... >> >> > >> >> > >> >> > thanks, >> >> > msp >> >> > >> >> >> >> -- >> >> View this message in context: >> >> >> http://www.nabble.com/Call-Variable-not-available-when-call-ha >> ngup-tp21788550p21789152.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com> ny.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> f%2B888 at conference.freeswitch.org> >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Call-Variable-not-available-when-call-ha >> ngup-tp21788550p21791503.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> ------------------------------ >> >> Message: 2 >> Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST) >> From: clif at eugeneweb.com >> Subject: Re: [Freeswitch-users] How do I set my FS internal ip address >> to a "static" value. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed >> >> Hi Gang, >> >> I've been struggleing with this also. Actually I can get it >> to bind to my >> address, the problem is it randomly drops my calls. :-( >> >> I have a FS running on a box with a static IP and I can start >> a call between >> two extensions and it will go for hours. Then I add anther >> interface say eth0:0 >> with a new static IP and reconfigure my phones and FS to use >> that, and the >> calls drop after about 15-20 mins. Though it's pretty random. >> >> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and >> freeswitch-1.0.1. >> Here is my /etc/network/interfaces: >> >> # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) >> >> # The loopback interface >> auto lo >> iface lo inet loopback >> >> # The first network card - this entry was created during the Debian >> installation >> auto eth0 eth0:0 >> iface eth0 inet dhcp >> iface eth0:0 inet static >> address 192.168.0.249 >> netmask 255.255.255.0 >> gateway 192.168.0.254 >> >> The only change I made to the FS config is in Vars.xml. I >> added this line close >> to the top: >> >> >> >> Here is the console log of the call being dropped: >> >> freeswitch at archive> sofia status >> API CALL [sofia(status)] output: >> Name Type >> Data >> State >> ============================================================== >> =================================== >> external profile >> sip:mod_sofia at 67.171.158.226:5080 >> RUNNING (0) >> internal profile >> sip:mod_sofia at 192.168.0.249:5060 >> RUNNING (2) >> nat profile >> sip:mod_sofia at 67.171.158.226:5070 >> RUNNING (0) >> default alias >> internal >> ALIASED >> outbound alias >> external >> ALIASED >> 192.168.0.249 alias >> internal >> ALIASED >> ============================================================== >> =================================== >> 3 profiles 3 aliases >> >> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 >> sofia_glue_restart_all_profiles() Reload XML [Success] >> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() >> ENUM Reloaded >> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 >> sofia_read_frame() Hangup >> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp >> ;fs_nat=yes >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 >> switch_ivr_multi_threaded_bridge() Hangup >> sofia/internal/1001 at 192.168.0.249 >> [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >> switch_core_session_thread() Session 6 >> (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud >> p;fs_nat=yes) >> Ended >> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >> switch_core_session_thread() Close Channel >> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp >> ;fs_nat=yes >> [CS_HANGUP] >> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >> switch_core_session_thread() Session 5 >> (sofia/internal/1001 at 192.168.0.249) >> Ended >> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >> switch_core_session_thread() Close Channel >> sofia/internal/1001 at 192.168.0.249 >> [CS_HANGUP] >> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() waiting for >> worker thread >> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() waiting for >> worker thread >> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias >> [192.168.0.249] for profile [internal] >> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() >> Adding Alias [default] >> for profile [internal] >> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() >> Started Profile >> internal [sofia_reg_internal] >> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias >> [outbound] for profile [external] >> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() >> Started Profile >> external [sofia_reg_external] >> 2009-02-01 13:23:20 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() waiting for >> worker thread >> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() >> Started Profile nat >> [sofia_reg_nat] >> sofia status >> API CALL [sofia(status)] output: >> Name Type >> Data >> State >> ============================================================== >> =================================== >> external profile >> sip:mod_sofia at 67.171.158.226:5080 >> RUNNING (0) >> internal profile >> sip:mod_sofia at 192.168.0.249:5060 >> RUNNING (0) >> outbound alias >> external >> ALIASED >> 192.168.0.249 alias >> internal >> ALIASED >> nat profile >> sip:mod_sofia at 67.171.158.226:5070 >> RUNNING (0) >> default alias >> internal >> ALIASED >> ============================================================== >> =================================== >> 3 profiles 3 aliases >> >> There is an older thread that says one should set >> >> but in this (later) thread is says only Jingleling usese that >> variable. >> ie. see: >> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >> org/msg00695.html >> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >> org/msg07345.html >> >> So what do you think causes this? What is the correct way? ;-) >> >> >> Thanks, >> Clif >> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Mon, 2 Feb 2009 09:41:05 -0600 >> From: Anthony Minessale >> Subject: Re: [Freeswitch-users] Call Variable not available when call >> hangup >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> you can't that's why i said it was a horrible approach. >> That's also why i posted you the instructions on the only >> elegant solution >> to your problem. >> >> >> On Mon, Feb 2, 2009 at 9:21 AM, shehzad p wrote: >> >> > >> > >> > one question is that when javascript is being called from >> dial plan, I get >> > the session object already available, It is for A leg of channel, >> > So when javascript is called after Bridge how can I get the >> session object >> > for B leg also? >> > >> > >> > Anthony Minessale-2 wrote: >> > > >> > > the leg you are running the script on is not hungup, the >> other leg of the >> > > call is. >> > > >> > > If it was hungup you would not be executing the script. >> > > >> > > Asterisk and the h ext and the whole dead-agi thing are >> all poor design >> > > showing it's teeth. >> > > We do not support anything like it. >> > > >> > > >> > > You can however try this: (see the link below) >> > > >> > > >> > >> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks >> -giving-me-headaches-p21614840.html >> > > >> > > >> > > >> > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p >> wrote: >> > > >> > >> >> > >> Is there any settings that when call hangup control can >> be transferred >> > to >> > >> another context and these CDR values can be accessible >> there? (just like >> > >> in >> > >> Asterisk, h extension) >> > >> >> > >> shehzad p wrote: >> > >> > >> > >> > Hi all, >> > >> > >> > >> > I need to process some CDR variables in Dialplan, like >> call duration, >> > >> > Answered time etc. >> > >> > but when I place info application after bridge, it is >> not listing them >> > >> > properly as below: >> > >> > =========================================== >> > >> > Caller-Channel-Created-Time: [1233573341672157] >> > >> > Caller-Channel-Answered-Time: [1233573342712939] >> > >> > Caller-Channel-Hangup-Time: [0] >> > >> > ========================================== >> > >> > Here Hangup time is 0, So how can I find actual values? >> > >> > >> > >> > --I know that we can use xml_cdr or cdr_csv, but my >> current need is to >> > >> get >> > >> > those values from dialplan itself so that can be passed to some >> > >> script... >> > >> > >> > >> > >> > >> > thanks, >> > >> > msp >> > >> > >> > >> >> > >> -- >> > >> View this message in context: >> > >> >> > >> http://www.nabble.com/Call-Variable-not-available-when-call-ha >> ngup-tp21788550p21789152.html >> > >> Sent from the Freeswitch-users mailing list archive at >> Nabble.com. >> > >> >> > >> >> > >> _______________________________________________ >> > >> Freeswitch-users mailing list >> > >> Freeswitch-users at lists.freeswitch.org >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE: >> > http://lists.freeswitch.org/mailman/options/freeswitch-users >> > >> http://www.freeswitch.org >> > >> >> > > >> > > >> > > >> > > -- >> > > Anthony Minessale II >> > > >> > > FreeSWITCH http://www.freeswitch.org/ >> > > ClueCon http://www.cluecon.com/ >> > > >> > > AIM: anthm >> > > MSN:anthony_minessale at hotmail.com >> < >> > >> MSN%3Aanthony_minessale at hotmail.com> hotmail.com> >> > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com> ny.minessale at gmail.com> >> > >> > sale at gmail.com> >> > > >> > > IRC: irc.freenode.net #freeswitch >> > > >> > > FreeSWITCH Developer Conference >> > > sip:888 at conference.freeswitch.org >> < >> > >> sip%3A888 at conference.freeswitch.org> eswitch.org> >> > > >> > > iax:guest at conference.freeswitch.org/888 >> > > >> googletalk:conf+888 at conference.freeswitch.org> f%2B888 at conference.freeswitch.org> >> > >> > 253Aconf%252B888 at conference.freeswitch.org> >> > > >> > > pstn:213-799-1400 >> > > >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > Freeswitch-users at lists.freeswitch.org >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> > > http://www.freeswitch.org >> > > >> > > >> > >> > -- >> > View this message in context: >> > >> http://www.nabble.com/Call-Variable-not-available-when-call-ha >> ngup-tp21788550p21791503.html >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com> ny.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> f%2B888 at conference.freeswitch.org> >> pstn:213-799-1400 >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachm >> ents/20090202/2d430e44/attachment-0001.html >> >> ------------------------------ >> >> Message: 4 >> Date: Mon, 2 Feb 2009 09:41:39 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] How do I set my FS internal ip address >> to a "static" value. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org> >> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes >> >> you need to add this setting to sofia.conf.xml >> >> >> >> >> You'll also need to edit the sofia profiles and input the >> exact IP you >> wish it to bind to. The params are sip-ip and rtp-ip. >> >> /b >> >> On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote: >> >> > Hi Gang, >> > >> > I've been struggleing with this also. Actually I can get it >> to bind >> > to my >> > address, the problem is it randomly drops my calls. :-( >> > >> > I have a FS running on a box with a static IP and I can >> start a call >> > between >> > two extensions and it will go for hours. Then I add anther >> interface >> > say eth0:0 >> > with a new static IP and reconfigure my phones and FS to use that, >> > and the >> > calls drop after about 15-20 mins. Though it's pretty random. >> > >> > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and >> > freeswitch-1.0.1. >> > Here is my /etc/network/interfaces: >> > >> > # /etc/network/interfaces -- configuration file for >> ifup(8), ifdown(8) >> > >> > # The loopback interface >> > auto lo >> > iface lo inet loopback >> > >> > # The first network card - this entry was created during the Debian >> > installation >> > auto eth0 eth0:0 >> > iface eth0 inet dhcp >> > iface eth0:0 inet static >> > address 192.168.0.249 >> > netmask 255.255.255.0 >> > gateway 192.168.0.254 >> > >> > The only change I made to the FS config is in Vars.xml. I >> added this >> > line close >> > to the top: >> > >> > >> > >> > Here is the console log of the call being dropped: >> > >> > freeswitch at archive> sofia status >> > API CALL [sofia(status)] output: >> > Name Type >> Data >> > State >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > >> ====================================================================== >> > external profile >> sip:mod_sofia at 67.171.158.226:5080 >> > RUNNING (0) >> > internal profile >> sip:mod_sofia at 192.168.0.249:5060 >> > RUNNING (2) >> > nat profile >> sip:mod_sofia at 67.171.158.226:5070 >> > RUNNING (0) >> > default alias >> internal >> > ALIASED >> > outbound alias >> external >> > ALIASED >> > 192.168.0.249 alias >> internal >> > ALIASED >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > >> ====================================================================== >> > 3 profiles 3 aliases >> > >> > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 >> > sofia_glue_restart_all_profiles() Reload XML [Success] >> > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM >> > Reloaded >> > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 >> sofia_read_frame() Hangup >> > sofia/internal/ >> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes >> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 >> > switch_ivr_multi_threaded_bridge() Hangup >> sofia/internal/1001 at 192.168.0.249 >> > [CS_EXECUTE] [NORMAL_CLEARING] >> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >> > switch_core_session_thread() Session 6 >> > (sofia/internal/ >> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) >> > Ended >> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >> > switch_core_session_thread() Close Channel >> > sofia/internal/ >> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes >> > [CS_HANGUP] >> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >> > switch_core_session_thread() Session 5 >> (sofia/internal/1001 at 192.168.0.249 >> > ) >> > Ended >> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >> > switch_core_session_thread() Close Channel >> sofia/internal/1001 at 192.168.0.249 >> > [CS_HANGUP] >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() >> > waiting for >> > worker thread >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() >> > waiting for >> > worker thread >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() >> Adding Alias >> > [192.168.0.249] for profile [internal] >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding >> > Alias [default] >> > for profile [internal] >> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started >> > Profile >> > internal [sofia_reg_internal] >> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() >> Adding Alias >> > [outbound] for profile [external] >> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started >> > Profile >> > external [sofia_reg_external] >> > 2009-02-01 13:23:20 [NOTICE] sofia.c:645 >> sofia_profile_thread_run() >> > waiting for >> > worker thread >> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started >> > Profile nat >> > [sofia_reg_nat] >> > sofia status >> > API CALL [sofia(status)] output: >> > Name Type >> Data >> > State >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > >> ====================================================================== >> > external profile >> sip:mod_sofia at 67.171.158.226:5080 >> > RUNNING (0) >> > internal profile >> sip:mod_sofia at 192.168.0.249:5060 >> > RUNNING (0) >> > outbound alias >> external >> > ALIASED >> > 192.168.0.249 alias >> internal >> > ALIASED >> > nat profile >> sip:mod_sofia at 67.171.158.226:5070 >> > RUNNING (0) >> > default alias >> internal >> > ALIASED >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > = >> > >> ====================================================================== >> > 3 profiles 3 aliases >> > >> > There is an older thread that says one should set >> > >> > but in this (later) thread is says only Jingleling usese that >> > variable. >> > ie. see: >> > >> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >> org/msg00695.html >> > >> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >> org/msg07345.html >> > >> > So what do you think causes this? What is the correct way? ;-) >> > >> > >> > Thanks, >> > Clif >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> > http://www.freeswitch.org >> >> >> >> >> ------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >> itch-users >> http://www.freeswitch.org >> >> >> End of Freeswitch-users Digest, Vol 32, Issue 17 >> ************************************************ >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From wojtek at VoIPMan.ORG Mon Mar 2 04:32:31 2009 From: wojtek at VoIPMan.ORG (Wojciech Tryc) Date: Mon, 2 Mar 2009 07:32:31 -0500 Subject: [Freeswitch-users] Running freeswitch on powerpc In-Reply-To: <7b197bef0903020411q72be83efxace263ab3401c001@mail.gmail.com> References: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> <7b197bef0903020411q72be83efxace263ab3401c001@mail.gmail.com> Message-ID: <60097A1C-2820-4397-BBEE-141FC7FEE3AE@VoIPMan.ORG> Sridhar, PIKA's WARP is PowerPC based...AMCC but still Big Endian and PowerPC. From what I remember the endianness definition was broken in one or two places, but other than that it was effortless (native compilation). Thanks, Wojtek, On Mar 2, 2009, at 7:11 AM, Giovanni Maruzzelli wrote: > On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar) > wrote: >> I am planning to run freeswitch on powerpc MPC8358. Please let me >> know if any changes needs to be done in the code > > Hi Sridhar, > > I don't think someone has tried that. It will probably be you that let > us all know which (if any) changes needs to be done. :-) > > > 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 Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar) > wrote: >> Hi all, >> >> I am planning to run freeswitch on powerpc MPC8358. Please let me >> know if any changes needs to be done in the code >> >> Regards >> Sridhar >> >> >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On >>> Behalf Of freeswitch-users-request at lists.freeswitch.org >>> Sent: Monday, February 02, 2009 9:12 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Freeswitch-users Digest, Vol 32, Issue 17 >>> >>> Send Freeswitch-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more >>> specific than "Re: Contents of Freeswitch-users digest..." >>> >>> >>> Today's Topics: >>> >>> 1. Re: Call Variable not available when call hangup (shehzad p) >>> 2. Re: How do I set my FS internal ip address to a "static" >>> value. (clif at eugeneweb.com) >>> 3. Re: Call Variable not available when call hangup >>> (Anthony Minessale) >>> 4. Re: How do I set my FS internal ip address to a "static" >>> value. (Brian West) >>> >>> >>> ---------------------------------------------------------------------- >>> >>> Message: 1 >>> Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST) >>> From: shehzad p >>> Subject: Re: [Freeswitch-users] Call Variable not available when >>> call >>> hangup >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: <21791503.post at talk.nabble.com> >>> Content-Type: text/plain; charset=us-ascii >>> >>> >>> >>> one question is that when javascript is being called from >>> dial plan, I get the session object already available, It is >>> for A leg of channel, So when javascript is called after >>> Bridge how can I get the session object for B leg also? >>> >>> >>> Anthony Minessale-2 wrote: >>>> >>>> the leg you are running the script on is not hungup, the >>> other leg of the >>>> call is. >>>> >>>> If it was hungup you would not be executing the script. >>>> >>>> Asterisk and the h ext and the whole dead-agi thing are all >>> poor design >>>> showing it's teeth. >>>> We do not support anything like it. >>>> >>>> >>>> You can however try this: (see the link below) >>>> >>>> >>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks >>> -giving-me-headaches-p21614840.html >>>> >>>> >>>> >>>> On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: >>>> >>>>> >>>>> Is there any settings that when call hangup control can be >>> transferred to >>>>> another context and these CDR values can be accessible >>> there? (just like >>>>> in >>>>> Asterisk, h extension) >>>>> >>>>> shehzad p wrote: >>>>>> >>>>>> Hi all, >>>>>> >>>>>> I need to process some CDR variables in Dialplan, like >>> call duration, >>>>>> Answered time etc. >>>>>> but when I place info application after bridge, it is >>> not listing them >>>>>> properly as below: >>>>>> =========================================== >>>>>> Caller-Channel-Created-Time: [1233573341672157] >>>>>> Caller-Channel-Answered-Time: [1233573342712939] >>>>>> Caller-Channel-Hangup-Time: [0] >>>>>> ========================================== >>>>>> Here Hangup time is 0, So how can I find actual values? >>>>>> >>>>>> --I know that we can use xml_cdr or cdr_csv, but my >>> current need is to >>>>> get >>>>>> those values from dialplan itself so that can be passed to some >>>>> script... >>>>>> >>>>>> >>>>>> thanks, >>>>>> msp >>>>>> >>>>> >>>>> -- >>>>> View this message in context: >>>>> >>> http://www.nabble.com/Call-Variable-not-available-when-call-ha >>> ngup-tp21788550p21789152.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>> >>>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>> ny.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>> f%2B888 at conference.freeswitch.org> >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Call-Variable-not-available-when-call-ha >>> ngup-tp21788550p21791503.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> >>> >>> ------------------------------ >>> >>> Message: 2 >>> Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST) >>> From: clif at eugeneweb.com >>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip >>> address >>> to a "static" value. >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: >>> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed >>> >>> Hi Gang, >>> >>> I've been struggleing with this also. Actually I can get it >>> to bind to my >>> address, the problem is it randomly drops my calls. :-( >>> >>> I have a FS running on a box with a static IP and I can start >>> a call between >>> two extensions and it will go for hours. Then I add anther >>> interface say eth0:0 >>> with a new static IP and reconfigure my phones and FS to use >>> that, and the >>> calls drop after about 15-20 mins. Though it's pretty random. >>> >>> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and >>> freeswitch-1.0.1. >>> Here is my /etc/network/interfaces: >>> >>> # /etc/network/interfaces -- configuration file for ifup(8), >>> ifdown(8) >>> >>> # The loopback interface >>> auto lo >>> iface lo inet loopback >>> >>> # The first network card - this entry was created during the Debian >>> installation >>> auto eth0 eth0:0 >>> iface eth0 inet dhcp >>> iface eth0:0 inet static >>> address 192.168.0.249 >>> netmask 255.255.255.0 >>> gateway 192.168.0.254 >>> >>> The only change I made to the FS config is in Vars.xml. I >>> added this line close >>> to the top: >>> >>> >>> >>> Here is the console log of the call being dropped: >>> >>> freeswitch at archive> sofia status >>> API CALL [sofia(status)] output: >>> Name Type >>> Data >>> State >>> ============================================================== >>> =================================== >>> external profile >>> sip:mod_sofia at 67.171.158.226:5080 >>> RUNNING (0) >>> internal profile >>> sip:mod_sofia at 192.168.0.249:5060 >>> RUNNING (2) >>> nat profile >>> sip:mod_sofia at 67.171.158.226:5070 >>> RUNNING (0) >>> default alias >>> internal >>> ALIASED >>> outbound alias >>> external >>> ALIASED >>> 192.168.0.249 alias >>> internal >>> ALIASED >>> ============================================================== >>> =================================== >>> 3 profiles 3 aliases >>> >>> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 >>> sofia_glue_restart_all_profiles() Reload XML [Success] >>> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() >>> ENUM Reloaded >>> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 >>> sofia_read_frame() Hangup >>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp >>> ;fs_nat=yes >>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 >>> switch_ivr_multi_threaded_bridge() Hangup >>> sofia/internal/1001 at 192.168.0.249 >>> [CS_EXECUTE] [NORMAL_CLEARING] >>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >>> switch_core_session_thread() Session 6 >>> (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud >>> p;fs_nat=yes) >>> Ended >>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >>> switch_core_session_thread() Close Channel >>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp >>> ;fs_nat=yes >>> [CS_HANGUP] >>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >>> switch_core_session_thread() Session 5 >>> (sofia/internal/1001 at 192.168.0.249) >>> Ended >>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >>> switch_core_session_thread() Close Channel >>> sofia/internal/1001 at 192.168.0.249 >>> [CS_HANGUP] >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() waiting for >>> worker thread >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() waiting for >>> worker thread >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding >>> Alias >>> [192.168.0.249] for profile [internal] >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() >>> Adding Alias [default] >>> for profile [internal] >>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() >>> Started Profile >>> internal [sofia_reg_internal] >>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding >>> Alias >>> [outbound] for profile [external] >>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() >>> Started Profile >>> external [sofia_reg_external] >>> 2009-02-01 13:23:20 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() waiting for >>> worker thread >>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() >>> Started Profile nat >>> [sofia_reg_nat] >>> sofia status >>> API CALL [sofia(status)] output: >>> Name Type >>> Data >>> State >>> ============================================================== >>> =================================== >>> external profile >>> sip:mod_sofia at 67.171.158.226:5080 >>> RUNNING (0) >>> internal profile >>> sip:mod_sofia at 192.168.0.249:5060 >>> RUNNING (0) >>> outbound alias >>> external >>> ALIASED >>> 192.168.0.249 alias >>> internal >>> ALIASED >>> nat profile >>> sip:mod_sofia at 67.171.158.226:5070 >>> RUNNING (0) >>> default alias >>> internal >>> ALIASED >>> ============================================================== >>> =================================== >>> 3 profiles 3 aliases >>> >>> There is an older thread that says one should set >>> >>> but in this (later) thread is says only Jingleling usese that >>> variable. >>> ie. see: >>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >>> org/msg00695.html >>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >>> org/msg07345.html >>> >>> So what do you think causes this? What is the correct way? ;-) >>> >>> >>> Thanks, >>> Clif >>> >>> >>> >>> >>> ------------------------------ >>> >>> Message: 3 >>> Date: Mon, 2 Feb 2009 09:41:05 -0600 >>> From: Anthony Minessale >>> Subject: Re: [Freeswitch-users] Call Variable not available when >>> call >>> hangup >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: >>> <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com> >>> Content-Type: text/plain; charset="iso-8859-1" >>> >>> you can't that's why i said it was a horrible approach. >>> That's also why i posted you the instructions on the only >>> elegant solution >>> to your problem. >>> >>> >>> On Mon, Feb 2, 2009 at 9:21 AM, shehzad p wrote: >>> >>>> >>>> >>>> one question is that when javascript is being called from >>> dial plan, I get >>>> the session object already available, It is for A leg of channel, >>>> So when javascript is called after Bridge how can I get the >>> session object >>>> for B leg also? >>>> >>>> >>>> Anthony Minessale-2 wrote: >>>>> >>>>> the leg you are running the script on is not hungup, the >>> other leg of the >>>>> call is. >>>>> >>>>> If it was hungup you would not be executing the script. >>>>> >>>>> Asterisk and the h ext and the whole dead-agi thing are >>> all poor design >>>>> showing it's teeth. >>>>> We do not support anything like it. >>>>> >>>>> >>>>> You can however try this: (see the link below) >>>>> >>>>> >>>> >>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks >>> -giving-me-headaches-p21614840.html >>>>> >>>>> >>>>> >>>>> On Mon, Feb 2, 2009 at 6:53 AM, shehzad p >>> wrote: >>>>> >>>>>> >>>>>> Is there any settings that when call hangup control can >>> be transferred >>>> to >>>>>> another context and these CDR values can be accessible >>> there? (just like >>>>>> in >>>>>> Asterisk, h extension) >>>>>> >>>>>> shehzad p wrote: >>>>>>> >>>>>>> Hi all, >>>>>>> >>>>>>> I need to process some CDR variables in Dialplan, like >>> call duration, >>>>>>> Answered time etc. >>>>>>> but when I place info application after bridge, it is >>> not listing them >>>>>>> properly as below: >>>>>>> =========================================== >>>>>>> Caller-Channel-Created-Time: [1233573341672157] >>>>>>> Caller-Channel-Answered-Time: [1233573342712939] >>>>>>> Caller-Channel-Hangup-Time: [0] >>>>>>> ========================================== >>>>>>> Here Hangup time is 0, So how can I find actual values? >>>>>>> >>>>>>> --I know that we can use xml_cdr or cdr_csv, but my >>> current need is to >>>>>> get >>>>>>> those values from dialplan itself so that can be passed to some >>>>>> script... >>>>>>> >>>>>>> >>>>>>> thanks, >>>>>>> msp >>>>>>> >>>>>> >>>>>> -- >>>>>> View this message in context: >>>>>> >>>> >>> http://www.nabble.com/Call-Variable-not-available-when-call-ha >>> ngup-tp21788550p21789152.html >>>>>> Sent from the Freeswitch-users mailing list archive at >>> Nabble.com. >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>> < >>>> >>> MSN%3Aanthony_minessale at hotmail.com>> hotmail.com> >>>>> >>>>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>> ny.minessale at gmail.com> >>>> >>> >> sale at gmail.com> >>>>> >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>> < >>>> >>> sip%3A888 at conference.freeswitch.org>> eswitch.org> >>>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>> googletalk:conf+888 at conference.freeswitch.org>> f%2B888 at conference.freeswitch.org> >>>> >>> >> 253Aconf%252B888 at conference.freeswitch.org> >>>>> >>>>> pstn:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> >>> http://www.nabble.com/Call-Variable-not-available-when-call-ha >>> ngup-tp21788550p21791503.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>> ny.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>> f%2B888 at conference.freeswitch.org> >>> pstn:213-799-1400 >>> -------------- next part -------------- >>> An HTML attachment was scrubbed... >>> URL: >>> http://lists.freeswitch.org/pipermail/freeswitch-users/attachm >>> ents/20090202/2d430e44/attachment-0001.html >>> >>> ------------------------------ >>> >>> Message: 4 >>> Date: Mon, 2 Feb 2009 09:41:39 -0600 >>> From: Brian West >>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip >>> address >>> to a "static" value. >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org> >>> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes >>> >>> you need to add this setting to sofia.conf.xml >>> >>> >>> >>> >>> You'll also need to edit the sofia profiles and input the >>> exact IP you >>> wish it to bind to. The params are sip-ip and rtp-ip. >>> >>> /b >>> >>> On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote: >>> >>>> Hi Gang, >>>> >>>> I've been struggleing with this also. Actually I can get it >>> to bind >>>> to my >>>> address, the problem is it randomly drops my calls. :-( >>>> >>>> I have a FS running on a box with a static IP and I can >>> start a call >>>> between >>>> two extensions and it will go for hours. Then I add anther >>> interface >>>> say eth0:0 >>>> with a new static IP and reconfigure my phones and FS to use that, >>>> and the >>>> calls drop after about 15-20 mins. Though it's pretty random. >>>> >>>> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and >>>> freeswitch-1.0.1. >>>> Here is my /etc/network/interfaces: >>>> >>>> # /etc/network/interfaces -- configuration file for >>> ifup(8), ifdown(8) >>>> >>>> # The loopback interface >>>> auto lo >>>> iface lo inet loopback >>>> >>>> # The first network card - this entry was created during the Debian >>>> installation >>>> auto eth0 eth0:0 >>>> iface eth0 inet dhcp >>>> iface eth0:0 inet static >>>> address 192.168.0.249 >>>> netmask 255.255.255.0 >>>> gateway 192.168.0.254 >>>> >>>> The only change I made to the FS config is in Vars.xml. I >>> added this >>>> line close >>>> to the top: >>>> >>>> >>>> >>>> Here is the console log of the call being dropped: >>>> >>>> freeswitch at archive> sofia status >>>> API CALL [sofia(status)] output: >>>> Name Type >>> Data >>>> State >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> >>> = >>> = >>> ==================================================================== >>>> external profile >>> sip:mod_sofia at 67.171.158.226:5080 >>>> RUNNING (0) >>>> internal profile >>> sip:mod_sofia at 192.168.0.249:5060 >>>> RUNNING (2) >>>> nat profile >>> sip:mod_sofia at 67.171.158.226:5070 >>>> RUNNING (0) >>>> default alias >>> internal >>>> ALIASED >>>> outbound alias >>> external >>>> ALIASED >>>> 192.168.0.249 alias >>> internal >>>> ALIASED >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> >>> = >>> = >>> ==================================================================== >>>> 3 profiles 3 aliases >>>> >>>> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 >>>> sofia_glue_restart_all_profiles() Reload XML [Success] >>>> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM >>>> Reloaded >>>> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 >>> sofia_read_frame() Hangup >>>> sofia/internal/ >>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes >>>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 >>>> switch_ivr_multi_threaded_bridge() Hangup >>> sofia/internal/1001 at 192.168.0.249 >>>> [CS_EXECUTE] [NORMAL_CLEARING] >>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >>>> switch_core_session_thread() Session 6 >>>> (sofia/internal/ >>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) >>>> Ended >>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >>>> switch_core_session_thread() Close Channel >>>> sofia/internal/ >>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes >>>> [CS_HANGUP] >>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 >>>> switch_core_session_thread() Session 5 >>> (sofia/internal/1001 at 192.168.0.249 >>>> ) >>>> Ended >>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 >>>> switch_core_session_thread() Close Channel >>> sofia/internal/1001 at 192.168.0.249 >>>> [CS_HANGUP] >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() >>>> waiting for >>>> worker thread >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() >>>> waiting for >>>> worker thread >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() >>> Adding Alias >>>> [192.168.0.249] for profile [internal] >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding >>>> Alias [default] >>>> for profile [internal] >>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started >>>> Profile >>>> internal [sofia_reg_internal] >>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() >>> Adding Alias >>>> [outbound] for profile [external] >>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started >>>> Profile >>>> external [sofia_reg_external] >>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:645 >>> sofia_profile_thread_run() >>>> waiting for >>>> worker thread >>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started >>>> Profile nat >>>> [sofia_reg_nat] >>>> sofia status >>>> API CALL [sofia(status)] output: >>>> Name Type >>> Data >>>> State >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> >>> = >>> = >>> ==================================================================== >>>> external profile >>> sip:mod_sofia at 67.171.158.226:5080 >>>> RUNNING (0) >>>> internal profile >>> sip:mod_sofia at 192.168.0.249:5060 >>>> RUNNING (0) >>>> outbound alias >>> external >>>> ALIASED >>>> 192.168.0.249 alias >>> internal >>>> ALIASED >>>> nat profile >>> sip:mod_sofia at 67.171.158.226:5070 >>>> RUNNING (0) >>>> default alias >>> internal >>>> ALIASED >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> = >>>> >>> = >>> = >>> ==================================================================== >>>> 3 profiles 3 aliases >>>> >>>> There is an older thread that says one should set >>>> >>>> but in this (later) thread is says only Jingleling usese that >>>> variable. >>>> ie. see: >>>> >>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >>> org/msg00695.html >>>> >>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch. >>> org/msg07345.html >>>> >>>> So what do you think causes this? What is the correct way? ;-) >>>> >>>> >>>> Thanks, >>>> Clif >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw >>> itch-users >>> http://www.freeswitch.org >>> >>> >>> End of Freeswitch-users Digest, Vol 32, Issue 17 >>> ************************************************ >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveu at coppice.org Mon Mar 2 04:50:29 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 02 Mar 2009 20:50:29 +0800 Subject: [Freeswitch-users] Running freeswitch on powerpc In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <49ABD615.9050906@coppice.org> Rajagopal, Sridhar (Sridhar) wrote: > Hi all, > > I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code > > Regards > Sridhar > It may be easier to say what will currently stop Freeswitch working. The lack of an MMU is a problem right now, so Blackfins are out, which is sad. Cores without hardware floating point may not perform all that well, but should work. Endianness should not be a problem. I think machines which choke on misaligned access are probably OK, too. Checking that list, you should be OK on a PPC. Steve From anthony.minessale at gmail.com Mon Mar 2 05:50:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 07:50:32 -0600 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <2ea4d47e0903020129y676bbc35n8727f8906237edce@mail.gmail.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <49A6E1DB.3070806@freeswitch.org> <1235740392995-2395557.post@n2.nabble.com> <2ea4d47e0903020129y676bbc35n8727f8906237edce@mail.gmail.com> Message-ID: <191c3a030903020550x44ee80e3tcf33b805c7c30d5e@mail.gmail.com> pardon? ESL is just a client library for event socket to make it easier to make event socket apps. ESL == Event Socket Library On Mon, Mar 2, 2009 at 3:29 AM, Gopal krishnan wrote: > Hi, > Actually what is the difference between ESL in FS 1.0.3 and event socket > in FS 1.0.2. Is the FS 1.0.3 ESL superior? > > On Fri, Feb 27, 2009 at 6:43 PM, Rex_Alex wrote: > >> Hi All, I did what you have all suggested. Now its working perfectly. >> Thanks a lot for all your assistance. Rex. >> >> Raymond Chandler wrote: >> and it will probably be a good idea to do make phpmod-install so that the >> .so and .php files gets into the correct place to be included -Ray Mathieu >> Rene wrote: > > You need your distro's php dev pakage. > On 26-Feb-09, at >> 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh && >> ./configure && make >> install and did Mathieu's suggestion but getting >> error as below, >> [root at server esl]# make phpmod make >> MYLIB="../libesl.a" >> SOLINK="-shared -Xlinker -x" >> >> CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >> -g >> -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >> -Werror >> -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> >> -Wmissing-prototypes" >> >> CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >> -DHAVE_EDITLINE >> -g -ggdb -I../../libs/libedit/src/ -fPIC" >> CXX_CFLAGS="" -C php make[1]: >> php-config: Command not found make[1]: >> Entering directory >> `/root/freeswitch-1.0.3/libs/esl/php' g++ >> >> -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >> -ggdb >> -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >> >> esl_wrap.cpp:717:18: error: zend.h: No such file or directory >> >> esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >> >> esl_wrap.cpp:719:17: error: php.h: No such file or directory >> >> esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >> >> esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >> >> directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >> >> scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >> >> ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >> >> ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >> error: >> expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >> variable or >> field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error: >> ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was >> not declared in this scope >> esl_wrap.cpp:793: error: expected >> primary-expression before ?void? >> esl_wrap.cpp:793: error: expected >> primary-expression before ?*? token >> esl_wrap.cpp:793: error: ?type? was >> not declared in this scope >> esl_wrap.cpp:793: error: expected >> primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer >> expression list treated as >> compound expression esl_wrap.cpp:793: error: >> expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 >> make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: >> *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i >> wrong? Thanks, Rex >> >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34 >> AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it.. >> Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh && >> ./configure && >> make install And then do Mathieu's suggestion? -MC >> >> _______________________________________________ Freeswitch-users >> mailing >> list Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>> >> http://www.freeswitch.org >> >> >> >> ------------------------------------------------------------------------ >> >> View this message in context: Re: ESL Wrapper >> >> Sent from the >> freeswitch-users mailing list archive >> at Nabble.com. >> >> _______________________________________________ >> Freeswitch-users mailing >> list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> View this message in context: Re: ESL Wrapper >> Sent from the freeswitch-users mailing list archiveat Nabble.com. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thank you with regards, > Gopal, > PeopleTech Systems Private Limited > www.peopletech.co.in > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/0bf9dec0/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 2 05:56:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 07:56:50 -0600 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <49AB004C.6090604@gmx.net> References: <49AB004C.6090604@gmx.net> Message-ID: <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> put origination_caller_id_number in the dial string of any call and you can set the caller id individually for that leg {origination_caller_id_number=1234} On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX wrote: > Hello, > > I have the following problem while providing callback (mod_eventsocket > is used): > 1) I want to call a certain destination number A with a suppressed > caller_id_number (this works fine with some vars in the origination string) > 2) The destination number A picks up the phone and enters a target > number B by DTMF > 3) freeswitch then forwards the call to target number B by DTMF and I > want to show the number A. I do this with uuid_setvar. The problem is > that it still shows unknown. > This is all with SIP. > > uuid_setvar however worked if I did not set the caller_id_number to > unknown. Per default this is then "00000000000" and can then be changed > with uuid_setvar to the number of A. > But if I set caller_id_number to unknown I can no longer change it to A. > > Any hint? > > Best regards > Peter > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/7d5ce07e/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 2 05:59:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 07:59:34 -0600 Subject: [Freeswitch-users] Orginate: getting status of call fail In-Reply-To: References: Message-ID: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com> The best way would be to add a few custom variables and add a secondary system that monitors the CDR data and uses the custom variables to identify what you want to do with the failed calls. On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I?ve been running a test script written in lua which now works very well > thanks to Anthony?s fix to stream file. > > > > Right now I?m using an event socket to initiate the call and passing the > name of the script along with originate thus: > > > > $dialstring = "originate > {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/Mygw/phonenum > '&lua(helloworld.lua )'"; > > $result = $obj ->bgapi_command($dialstring); > > > > The script gets fired (it would appear) on answer. However, if the number > is invalid , timed out or was busy, I?m not sure the script gets executed or > am I wrong? > > > > I want to be able to fire an event back on what happed to the call in the > event that it failed for whatever reason. > > > > I know I can simply call the originate and pass the number as an argument > and execute the dial within the script but I?m led to believe that?s not > very efficient, or am I completely wrong? > > > > Looking for the most FS friendly way here > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/d35279cc/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Mar 2 06:49:03 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 2 Mar 2009 14:49:03 -0000 Subject: [Freeswitch-users] Orginate: getting status of call fail In-Reply-To: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com> References: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com> Message-ID: That's what I was wondering, however, won't the response to the bagi (not the initial) give me the info on the call result? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 02 March 2009 14:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Orginate: getting status of call fail The best way would be to add a few custom variables and add a secondary system that monitors the CDR data and uses the custom variables to identify what you want to do with the failed calls. On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton wrote: Hi Guys, I've been running a test script written in lua which now works very well thanks to Anthony's fix to stream file. Right now I'm using an event socket to initiate the call and passing the name of the script along with originate thus: $dialstring = "originate {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/ Mygw/phonenum '&lua(helloworld.lua )'"; $result = $obj ->bgapi_command($dialstring); The script gets fired (it would appear) on answer. However, if the number is invalid , timed out or was busy, I'm not sure the script gets executed or am I wrong? I want to be able to fire an event back on what happed to the call in the event that it failed for whatever reason. I know I can simply call the originate and pass the number as an argument and execute the dial within the script but I'm led to believe that's not very efficient, or am I completely wrong? Looking for the most FS friendly way here Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/2595216a/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 2 07:26:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 09:26:57 -0600 Subject: [Freeswitch-users] Orginate: getting status of call fail In-Reply-To: References: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com> Message-ID: <191c3a030903020726y455b786dka15206f2be5a7559@mail.gmail.com> yes if you match the job uuid from bgapi to the SWITCH_EVENT_BACKGROUND_JOB event, you would get the result in that event. On Mon, Mar 2, 2009 at 8:49 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > That?s what I was wondering, however, won?t the response to the bagi (not > the initial) give me the info on the call result? > > > > Regards > > > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 02 March 2009 14:00 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Orginate: getting status of call fail > > > > The best way would be to add a few custom variables and add a secondary > system that monitors the CDR data and uses the > custom variables to identify what you want to do with the failed calls. > > > On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Hi Guys, > > > > I?ve been running a test script written in lua which now works very well > thanks to Anthony?s fix to stream file. > > > > Right now I?m using an event socket to initiate the call and passing the > name of the script along with originate thus: > > > > $dialstring = "originate > {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/Mygw/phonenum > '&lua(helloworld.lua )'"; > > $result = $obj ->bgapi_command($dialstring); > > > > The script gets fired (it would appear) on answer. However, if the number > is invalid , timed out or was busy, I?m not sure the script gets executed or > am I wrong? > > > > I want to be able to fire an event back on what happed to the call in the > event that it failed for whatever reason. > > > > I know I can simply call the originate and pass the number as an argument > and execute the dial within the script but I?m led to believe that?s not > very efficient, or am I completely wrong? > > > > Looking for the most FS friendly way here > > > > Regards, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/f3e9cdd0/attachment-0002.html From a.playful.idiot at gmail.com Sun Mar 1 23:43:34 2009 From: a.playful.idiot at gmail.com (Aplayful Idiot) Date: Sun, 1 Mar 2009 23:43:34 -0800 Subject: [Freeswitch-users] First time test set up FreeSwitch and SPA3102/SPA3000 Message-ID: <9ed22e920903012343u5e50c89fkfa9d1127894199c@mail.gmail.com> I have no background in telephony but probably need to use a PBX. FreeSwitch was recommended by a casual contact so I would like to start first by setting up a small test. I have a SPA3102 attached to the box running FS and to a ordinary phone line. I registered SPA in conf/directory/default/line1.xml and it works ok but I can't get caller id numbers from incoming calls. All FS sees is "line1" which is found in file line1.xml as . Looking back over the FS wiki, I'm now wondering if the SPA should of been set-up as a gateway but reading it is confusing at least to me. Sometimes I think the analogue-line-SPA-FS is like a softphone which is registered to an extension numbered xml file in conf/directory/default/ but then issues like not getting outside incoming caller id's makes me think I've got this all wrong. Can someone help me out with this? api -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090301/eb6ea6e5/attachment-0002.html From a.playful.idiot at gmail.com Mon Mar 2 07:41:31 2009 From: a.playful.idiot at gmail.com (Aplayful Idiot) Date: Mon, 2 Mar 2009 07:41:31 -0800 Subject: [Freeswitch-users] First time setting up FreeSwitch and SPA3102 / SPA3000 Message-ID: <9ed22e920903020741o2ff66970h35d3182655b2c7ba@mail.gmail.com> Hi. I have little background in telephony and need to use a PBX but would like to start first with a small test set-up. I have a SPA3102 attached to the box running FS and to a ordinary phone line. I registered SPA in conf/directory/default/line1.xml and this works to a point but I can't get caller id numbers from incoming calls. All FS sees is "line1" which is found in file line1.xml as . Looking back over the FS wiki, I'm now wondering if the SPA was registered or set-up in FS correctly but reading the documentation is confusing me a bit. Sometimes I think the analogue-phone-line-SPA-FS is like a softphone which is registered to an extension numbered xml file in conf/directory/default/ but then issues like not getting outside incoming caller id's makes me think I've got this all wrong. Can someone help me out with this? Thanks. api -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/e073e954/attachment-0002.html From mike at jerris.com Mon Mar 2 08:14:52 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Mar 2009 11:14:52 -0500 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: <49AB8562.4050806@laposte.net> References: <49AB8562.4050806@laposte.net> Message-ID: Could you please post this to jira along with a thread apply all bt of a core file taken from the process with the stuck sessions. Mike On Mar 2, 2009, at 2:06 AM, rod wrote: > Hi All, > > I ran some longer tests with FS 1.0.3 acting as an SBC. > The test machine has the following specs: > - Intel Quad Core Q9550 > - 8GB RAM (far too much from what I saw) > > After 3 days running SIPP with 750 simultaneous calls (1500 > channels) at > 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. > > In the CLI, using status command I got this: > > freeswitch at internal> status > UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds, > 607 microseconds > 15817560 session(s) since startup > 22 session(s) 0/500 > > But when I use "show channels" or "show calls", I see nothing. So I'm > wondering where are these 22 sessions ? > > FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs. > > Successful call --> 5271434 > Failed call ---> 1554 (less than 0.03%) > > regards, > rod. > > > > complete SIPP summary: > > ------------------------------ Scenario Screen -------- [1-9]: Change > Screen -- > Call-rate(length) Port Total-time Total-calls Remote-host > 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 > 10.10.10.254:5060(UDP) > > 0 new calls during 0.856 s period 7 ms scheduler resolution > 0 calls (limit 750) Peak was 750 calls, after 15 s > 0 Running, 34 Paused, 0 Woken up > 15544 out-of-call msg (discarded) > 1 open sockets > 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate > (kB/s) > 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate > (kB/s) > > Messages Retrans Timeout > Unexpected-Msg > INVITE ----------> 5273022 0 0 > 100 <---------- 5273022 0 1554 > 180 <---------- 0 0 0 > 183 <---------- 0 0 0 > 200 <---------- E-RTD1 5271434 0 0 > ACK ----------> 5271434 0 > Pause [ 35.0s] 5271434 0 > BYE ----------> 5271434 0 0 > 200 <---------- 5271434 0 0 > > ------------------------------ Test Terminated > -------------------------------- > > > ----------------------------- Statistics Screen ------- [1-9]: Change > Screen -- > Start Time | 2009-02-27 > 09:11:31 > Last Reset Time | 2009-03-02 > 07:49:10 > Current Time | 2009-03-02 > 07:49:11 > -------------------------+--------------------------- > +-------------------------- > Counter Name | Periodic value | Cumulative value > -------------------------+--------------------------- > +-------------------------- > Elapsed Time | 00:00:00:857 | > 70:37:39:429 > Call Rate | 0.000 cps | 20.739 > cps > -------------------------+--------------------------- > +-------------------------- > Incoming call created | 0 | > 0 > OutGoing call created | 0 | > 5273022 > Total Call created | | > 5273022 > Current Call | 34 > | > -------------------------+--------------------------- > +-------------------------- > Successful call | 0 | > 5271434 > Failed call | 0 | > 1554 > -------------------------+--------------------------- > +-------------------------- > Response Time 1 | 00:00:00:000 | > 00:00:00:240 > Call Length | 38:32:13:386 | > 00:00:36:131 > ------------------------------ Test Terminated > -------------------------------- From mike at jerris.com Mon Mar 2 08:18:42 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Mar 2009 11:18:42 -0500 Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook? In-Reply-To: <7.0.1.0.2.20090302124612.028657a8@free.fr> References: <7.0.1.0.2.20090302124612.028657a8@free.fr> Message-ID: <485E559F-4D98-48AB-8CF0-EF3FD50DBB5F@jerris.com> I think any issues we have are related to pri, the analog doesn't seem to generate any major bug reports. Mike On Mar 2, 2009, at 6:47 AM, Fred wrote: > Thanks guys for the feedback. So, the OpenZap driver isn't ready for > production yet? > From Prometheus001 at gmx.net Mon Mar 2 08:19:32 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 02 Mar 2009 17:19:32 +0100 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> Message-ID: <49AC0714.8000006@gmx.net> Hello Anthony, I do this when I orginate the call. This way we suppress the cid when we call party A and transfer A to an internal extension (our callback application). But now comes the part that does not work: After A enters the target number B (via DTMF), we set the cid variables via uuid_setvar and then transfer A via uuid_transfer to party B. However uuid_setvar does not work in that case. BUT: If we do the same scenario and do not suppress the cid in the originate part, then uuid_setvar works correctly and sets the cid_number. Best regards Peter Anthony Minessale schrieb: > put origination_caller_id_number in the dial string of any call and > you can set the caller id individually for that leg > > {origination_caller_id_number=1234} > > > On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX > wrote: > > Hello, > > I have the following problem while providing callback (mod_eventsocket > is used): > 1) I want to call a certain destination number A with a suppressed > caller_id_number (this works fine with some vars in the > origination string) > 2) The destination number A picks up the phone and enters a target > number B by DTMF > 3) freeswitch then forwards the call to target number B by DTMF and I > want to show the number A. I do this with uuid_setvar. The problem is > that it still shows unknown. > This is all with SIP. > > uuid_setvar however worked if I did not set the caller_id_number to > unknown. Per default this is then "00000000000" and can then be > changed > with uuid_setvar to the number of A. > But if I set caller_id_number to unknown I can no longer change it > to A. > > Any hint? > > Best regards > Peter > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From woof at nortel.com Mon Mar 2 09:16:33 2009 From: woof at nortel.com (Andy Spitzer) Date: Mon, 02 Mar 2009 12:16:33 -0500 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: Woof! On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote: > NO. You want something that people THINK exists and works well... > Reliable human/voice detection doesn't exist in ANY form. I beg to differ. See http://www.freepatentsonline.com/5521967.html for one way to do it. It works rather well and can quickly descriminate between voice and tone. I've no idea who owns that patent now (not me, for sure). There is a simpler, less reliable way of differentiating voice from tone, that as far as I know isn't patented. If you compare the RMS power levels of sequential 40 mS periods, call progress tones will have very consistent power levels from sample to sample. So if 5 or more 40 mS periods have about the same power measurement (within say, 2%), it's a tone. Voice will have dramatic power level differences over that same period. This works very well in today's telephony environment, where tones are computer generated. In the old days when ringback tone was generated off the audio hum from the 20 Hz ring voltage generator...not so well. --Woof! From msc at freeswitch.org Mon Mar 2 10:05:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Mar 2009 10:05:43 -0800 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49A7F393.6080406@ewetel.de> References: <49A7F393.6080406@ewetel.de> Message-ID: <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> On Fri, Feb 27, 2009 at 6:07 AM, Helmut Kuper wrote: > Hello, > > I play around with record_session and would like to have caller and > callee separated on left and right channel. I found record_stereo is > used for this. Unfortunately it doesn't work. A and B leg are still > mixed. Additionally I found that B leg is significant louder than A leg, > but both legs were local extensions. > Just to confirm - you are trying to record each leg of the call into a separate file? In other words, one call creates two separate audio recordings? -MC From anthony.minessale at gmail.com Mon Mar 2 11:48:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 13:48:26 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> i think that's what mod_vmd does On Mon, Mar 2, 2009 at 11:16 AM, Andy Spitzer wrote: > Woof! > > On Sun, 01 Mar 2009 21:28:18 -0500, Brian West > wrote: > > > NO. You want something that people THINK exists and works well... > > Reliable human/voice detection doesn't exist in ANY form. > > I beg to differ. See http://www.freepatentsonline.com/5521967.html for > one way to do it. It works rather well and can quickly descriminate between > voice and tone. I've no idea who owns that patent now (not me, for sure). > > There is a simpler, less reliable way of differentiating voice from tone, > that as far as I know isn't patented. If you compare the RMS power levels > of sequential 40 mS periods, call progress tones will have very consistent > power levels from sample to sample. So if 5 or more 40 mS periods have > about the same power measurement (within say, 2%), it's a tone. Voice will > have dramatic power level differences over that same period. This works > very well in today's telephony environment, where tones are computer > generated. In the old days when ringback tone was generated off the audio > hum from the 20 Hz ring voltage generator...not so well. > > --Woof! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/a8783b8b/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 2 11:52:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 13:52:16 -0600 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <49AC0714.8000006@gmx.net> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> <49AC0714.8000006@gmx.net> Message-ID: <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> origination_caller_id number is not ok as a variable unless its in {} as part of the dial string it's an exception that is parsed before the channel is even created. I think you are drawing the wrong conclusion about what works and doesn't work. If you can produce a dial string that contains {origination_caller_id_number=x} you will always be able to set it. I assume you are using a recent version of FS as we did have a small bug with this variable a few weeks ago. On Mon, Mar 2, 2009 at 10:19 AM, Peter P GMX wrote: > Hello Anthony, > > I do this when I orginate the call. This way we suppress the cid when we > call party A and transfer A to an internal extension (our callback > application). > But now comes the part that does not work: > After A enters the target number B (via DTMF), we set the cid variables > via uuid_setvar and then transfer A via uuid_transfer to party B. > However uuid_setvar does not work in that case. > > BUT: If we do the same scenario and do not suppress the cid in the > originate part, then uuid_setvar works correctly and sets the cid_number. > > Best regards > Peter > > Anthony Minessale schrieb: > > put origination_caller_id_number in the dial string of any call and > > you can set the caller id individually for that leg > > > > {origination_caller_id_number=1234} > > > > > > On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX > > wrote: > > > > Hello, > > > > I have the following problem while providing callback > (mod_eventsocket > > is used): > > 1) I want to call a certain destination number A with a suppressed > > caller_id_number (this works fine with some vars in the > > origination string) > > 2) The destination number A picks up the phone and enters a target > > number B by DTMF > > 3) freeswitch then forwards the call to target number B by DTMF and I > > want to show the number A. I do this with uuid_setvar. The problem is > > that it still shows unknown. > > This is all with SIP. > > > > uuid_setvar however worked if I did not set the caller_id_number to > > unknown. Per default this is then "00000000000" and can then be > > changed > > with uuid_setvar to the number of A. > > But if I set caller_id_number to unknown I can no longer change it > > to A. > > > > Any hint? > > > > Best regards > > Peter > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/21522899/attachment-0002.html From msc at freeswitch.org Mon Mar 2 12:03:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Mar 2009 12:03:52 -0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> Message-ID: <87f2f3b90903021203y3588ffa4nc3ccd280117e0129@mail.gmail.com> On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale wrote: > i think that's what mod_vmd does > I think that's right. It just does the opposite - instead of looking for differing power levels it looks for the same power level. In other words it tries to detect distinctly non-human sound. I'll bet you could futz with that code and have it fire off events when it detects what it believes is human speech. -MC From Prometheus001 at gmx.net Mon Mar 2 12:31:47 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 02 Mar 2009 21:31:47 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AB0BCD.8030108@gmx.net> References: <49A92BAE.4090907@gmx.net> <49AB0BCD.8030108@gmx.net> Message-ID: <49AC4233.6060506@gmx.net> Some more info: the system I am working on is a copy (dd copy) of a system where the pizza demo works on. The only thing I changed was to update to the current freeswitch trunk 12293 (it was 10003 before). Do I need to update the model? I did a make in the model directory, but no change. Best regards Peter Peter P GMX schrieb: > Hello Brian, > > thanks for the info. I am a step further, but it cannot load the grammar > files. > I am sending through event_socket: > > SendMsg > call-command: execute > execute-app-name: detect_speech > execute-app-arg: pocketsphinx yes no > > However I get the message (also when I am using Pizza demo): > 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() > sofia/internal/1000 at sip2.server.com Command Execute > detect_speech(pocketsphinx yes no) > 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 > pocketsphinx_asr_load_grammar() Can't open language model > /usr/local/freeswitch/grammar/model/communicator. > 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 > switch_ivr_detect_speech() Error loading Grammar > 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 > pocketsphinx_asr_close() Port Closed. > > However the grammar files are there: > root at sip2:/usr/local/freeswitch/grammar/model/communicator# > root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al > total 12752 > drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . > drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. > -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING > -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params > -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef > -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means > -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict > -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump > -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices > -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances > > > Any hint? > > Best regards > Peter > > Brian West schrieb: > >> You can accomplish this .... here is an example using ESL in perl >> >> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 >> >> /b >> >> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: >> >> >> >>> Or back to the basics: Is it possible to use pocketsphinx through >>> event >>> socket? >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From sergio.alecha at gmail.com Mon Mar 2 12:18:06 2009 From: sergio.alecha at gmail.com (Sergio Alecha) Date: Mon, 2 Mar 2009 17:18:06 -0300 Subject: [Freeswitch-users] Howto config early dial Message-ID: <47f9a0940903021218l1a6f7ea1l8c596ddd3514e86c@mail.gmail.com> In asterisk, with the parameter AMPBADNUMBER = FALSE it is possible to use "early dial" Grandstream telephones. How do Freeswitch in? thank you very much. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/beb261ca/attachment-0002.html From Prometheus001 at gmx.net Mon Mar 2 13:58:35 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 02 Mar 2009 22:58:35 +0100 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> <49AC0714.8000006@gmx.net> <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> Message-ID: <49AC568B.7020504@gmx.net> Hello Anthony, sorry for being tenacious but in some cases it works in a way we need it: If I a am not suppressing the cid numer when calling A, the following scenario works: * A receives a Call (originate) with CID '0000000000' (default from switch_caller.c) * A dials some digits via DTMF, the app set the cid variables via uuid_setvar and uuid_transfers the call to B. B receives a call with the right cid set. Maybe I simply modify the default cid '0000000000' to a different value in switch_caller.c? Is there a special reason why this is '0000000000'? I am using trunk version 12293. Best regards Peter Anthony Minessale schrieb: > origination_caller_id number is not ok as a variable unless its in {} > as part of the dial string > it's an exception that is parsed before the channel is even created. > > I think you are drawing the wrong conclusion about what works and > doesn't work. > If you can produce a dial string that contains > {origination_caller_id_number=x} you will always be able to set it. > > I assume you are using a recent version of FS as we did have a small > bug with this variable a few weeks ago. > > > On Mon, Mar 2, 2009 at 10:19 AM, Peter P GMX > wrote: > > Hello Anthony, > > I do this when I orginate the call. This way we suppress the cid > when we > call party A and transfer A to an internal extension (our callback > application). > But now comes the part that does not work: > After A enters the target number B (via DTMF), we set the cid > variables > via uuid_setvar and then transfer A via uuid_transfer to party B. > However uuid_setvar does not work in that case. > > BUT: If we do the same scenario and do not suppress the cid in the > originate part, then uuid_setvar works correctly and sets the > cid_number. > > Best regards > Peter > > Anthony Minessale schrieb: > > put origination_caller_id_number in the dial string of any call and > > you can set the caller id individually for that leg > > > > {origination_caller_id_number=1234} > > > > > > On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX > > > >> > wrote: > > > > Hello, > > > > I have the following problem while providing callback > (mod_eventsocket > > is used): > > 1) I want to call a certain destination number A with a > suppressed > > caller_id_number (this works fine with some vars in the > > origination string) > > 2) The destination number A picks up the phone and enters a > target > > number B by DTMF > > 3) freeswitch then forwards the call to target number B by > DTMF and I > > want to show the number A. I do this with uuid_setvar. The > problem is > > that it still shows unknown. > > This is all with SIP. > > > > uuid_setvar however worked if I did not set the > caller_id_number to > > unknown. Per default this is then "00000000000" and can then be > > changed > > with uuid_setvar to the number of A. > > But if I set caller_id_number to unknown I can no longer > change it > > to A. > > > > Any hint? > > > > Best regards > > Peter > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Mar 2 14:08:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Mar 2009 14:08:53 -0800 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <49AC568B.7020504@gmx.net> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> <49AC0714.8000006@gmx.net> <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> <49AC568B.7020504@gmx.net> Message-ID: <87f2f3b90903021408g62ba6a4eo925b70beb1a562a5@mail.gmail.com> On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX wrote: > Hello Anthony, > > sorry for being tenacious but in some cases it works in a way we need it: > If I a am not suppressing the cid numer when calling A, the following > scenario works: > > ? ?* A receives a Call (originate) with CID '0000000000' (default from > ? ? ?switch_caller.c) > ? ?* A dials some digits via DTMF, the app set the cid variables via > ? ? ?uuid_setvar and uuid_transfers the call to B. B receives a call > ? ? ?with the right cid set. > > Maybe I simply modify the default cid '0000000000' ?to a different value > in switch_caller.c? Is there a special reason why this is '0000000000'? > Check vars.xml to confirm that you have actually set a default caller ID. Most likely you'll still have the default caller id number set to all zeroes, which is the default. -MC > I am using trunk version 12293. > > Best regards > Peter > From anthony.minessale at gmail.com Mon Mar 2 14:22:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Mar 2009 16:22:25 -0600 Subject: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number In-Reply-To: <87f2f3b90903021408g62ba6a4eo925b70beb1a562a5@mail.gmail.com> References: <49AB004C.6090604@gmx.net> <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com> <49AC0714.8000006@gmx.net> <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com> <49AC568B.7020504@gmx.net> <87f2f3b90903021408g62ba6a4eo925b70beb1a562a5@mail.gmail.com> Message-ID: <191c3a030903021422i385ab5c0h781993c65d6d27b4@mail.gmail.com> Since you did not describe the exact way you are doing it with enough detail or any trace I can't begin to tell you what your problem is. you did not even mention what variable you are using or show examples. All I can do is tell you again that if you set the origination_caller_id_number in the dial string it will be the most likely to work for you. On Mon, Mar 2, 2009 at 4:08 PM, Michael Collins wrote: > On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX wrote: > > Hello Anthony, > > > > sorry for being tenacious but in some cases it works in a way we need it: > > If I a am not suppressing the cid numer when calling A, the following > > scenario works: > > > > * A receives a Call (originate) with CID '0000000000' (default from > > switch_caller.c) > > * A dials some digits via DTMF, the app set the cid variables via > > uuid_setvar and uuid_transfers the call to B. B receives a call > > with the right cid set. > > > > Maybe I simply modify the default cid '0000000000' to a different value > > in switch_caller.c? Is there a special reason why this is '0000000000'? > > > > Check vars.xml to confirm that you have actually set a default caller > ID. Most likely you'll still have the default caller id number set to > all zeroes, which is the default. > > -MC > > > I am using trunk version 12293. > > > > Best regards > > Peter > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/58b8ff69/attachment-0002.html From freeswitch at servercorps.com Mon Mar 2 14:43:04 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Mon, 2 Mar 2009 16:43:04 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AC4233.6060506@gmx.net> References: <49A92BAE.4090907@gmx.net> <49AB0BCD.8030108@gmx.net> <49AC4233.6060506@gmx.net> Message-ID: <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com> Peter, You need the grammar files for the pizza demo: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo has lonks to premade fles for everyhting to get the pizza demo working with pocketshinx. Those to not come with the source code when you update from SVN. Nik On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX wrote: > Some more info: > the system I am working on is a copy (dd copy) of a system where the > pizza demo works on. > The only thing I changed was to update to the current freeswitch trunk > 12293 (it was 10003 before). > > Do I need to update the model? I did a make in the model directory, but > no change. > > Best regards > Peter > > Peter P GMX schrieb: >> Hello Brian, >> >> thanks for the info. I am a step further, but it cannot load the grammar >> files. >> I am sending through event_socket: >> >> SendMsg >> call-command: execute >> execute-app-name: detect_speech >> execute-app-arg: pocketsphinx yes no >> >> However I get the message (also when I am using Pizza demo): >> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() >> sofia/internal/1000 at sip2.server.com Command Execute >> detect_speech(pocketsphinx yes no) >> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 >> pocketsphinx_asr_load_grammar() Can't open language model >> /usr/local/freeswitch/grammar/model/communicator. >> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 >> switch_ivr_detect_speech() Error loading Grammar >> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 >> pocketsphinx_asr_close() Port Closed. >> >> However the grammar files are there: >> root at sip2:/usr/local/freeswitch/grammar/model/communicator# >> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al >> total 12752 >> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . >> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. >> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING >> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params >> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef >> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means >> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict >> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump >> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices >> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances >> >> >> Any hint? >> >> Best regards >> Peter >> >> Brian West schrieb: >> >>> You can accomplish this .... here is an example using ESL in perl >>> >>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 >>> >>> /b >>> >>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: >>> >>> >>> >>>> Or back to the basics: Is it possible to use pocketsphinx through >>>> event >>>> socket? >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Mon Mar 2 15:42:28 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 03 Mar 2009 00:42:28 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com> References: <49A92BAE.4090907@gmx.net> <49AB0BCD.8030108@gmx.net> <49AC4233.6060506@gmx.net> <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com> Message-ID: <49AC6EE4.9080509@gmx.net> Thanks Addison. The Pizza files are there (as mentionned is it a copy of an already working system). In fact freeswitch is complaning about /usr/local/freeswitch/grammar/model/communicator which he cannot load So somehow freeswitch is not willing to open the files, but I have no clue why. So any hints are welcome. Best regards Peter Addison Martin schrieb: > Peter, > > You need the grammar files for the pizza demo: > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo > has lonks to premade fles for everyhting to get the pizza demo working > with pocketshinx. Those to not come with the source code when you > update from SVN. > > Nik > > > > On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX wrote: > >> Some more info: >> the system I am working on is a copy (dd copy) of a system where the >> pizza demo works on. >> The only thing I changed was to update to the current freeswitch trunk >> 12293 (it was 10003 before). >> >> Do I need to update the model? I did a make in the model directory, but >> no change. >> >> Best regards >> Peter >> >> Peter P GMX schrieb: >> >>> Hello Brian, >>> >>> thanks for the info. I am a step further, but it cannot load the grammar >>> files. >>> I am sending through event_socket: >>> >>> SendMsg >>> call-command: execute >>> execute-app-name: detect_speech >>> execute-app-arg: pocketsphinx yes no >>> >>> However I get the message (also when I am using Pizza demo): >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() >>> sofia/internal/1000 at sip2.server.com Command Execute >>> detect_speech(pocketsphinx yes no) >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 >>> pocketsphinx_asr_load_grammar() Can't open language model >>> /usr/local/freeswitch/grammar/model/communicator. >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 >>> switch_ivr_detect_speech() Error loading Grammar >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 >>> pocketsphinx_asr_close() Port Closed. >>> >>> However the grammar files are there: >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al >>> total 12752 >>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . >>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. >>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING >>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params >>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means >>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict >>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump >>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances >>> >>> >>> Any hint? >>> >>> Best regards >>> Peter >>> >>> Brian West schrieb: >>> >>> >>>> You can accomplish this .... here is an example using ESL in perl >>>> >>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 >>>> >>>> /b >>>> >>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: >>>> >>>> >>>> >>>> >>>>> Or back to the basics: Is it possible to use pocketsphinx through >>>>> event >>>>> socket? >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steveu at coppice.org Mon Mar 2 16:08:27 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 03 Mar 2009 08:08:27 +0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: <49AC74FB.3030903@coppice.org> Andy Spitzer wrote: > Woof! > > On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote: > > >> NO. You want something that people THINK exists and works well... >> Reliable human/voice detection doesn't exist in ANY form. >> > > I beg to differ. See http://www.freepatentsonline.com/5521967.html for one way to do it. It works rather well and can quickly descriminate between voice and tone. I've no idea who owns that patent now (not me, for sure). > Since when did a patent mean a problem is solved? For things like speech recognition you can achieve pretty high accuracy in voice detection, but in that case you can delay the audio and make decisions that span the start of the speech burst. For most telephony purposes you need to make a decision on the very first frame of speech, as you can't afford to add latency. That turns it into a tough problem. Something like the VAD in G.729 is about the best people can currently do, but its far from perfect. > There is a simpler, less reliable way of differentiating voice from tone, that as far as I know isn't patented. If you compare the RMS power levels of sequential 40 mS periods, call progress tones will have very consistent power levels from sample to sample. So if 5 or more 40 mS periods have about the same power measurement (within say, 2%), it's a tone. Voice will have dramatic power level differences over that same period. This works very well in today's telephony environment, where tones are computer generated. In the old days when ringback tone was generated off the audio hum from the 20 Hz ring voltage generator...not so well. > That is *not* VAD. What you describe just says "is its energy steady". I will trigger on music, background noise and maybe even some of the fast pulsed tone signals. A proper VAD won't. Regards, Steve From steveu at coppice.org Mon Mar 2 16:10:50 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 03 Mar 2009 08:10:50 +0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com> Message-ID: <49AC758A.60304@coppice.org> Hi, mod_vmd is a bit more sophisticated than that. It looks for the signal being narrowband energy. However, mod_vmd isn't very reliable, as it takes a rather high SNR for its narrowband detector to work. So high that a lossy codec like G.711 can barely manage it. Regards, Steve Anthony Minessale wrote: > i think that's what mod_vmd does > > On Mon, Mar 2, 2009 at 11:16 AM, Andy Spitzer > wrote: > > Woof! > > On Sun, 01 Mar 2009 21:28:18 -0500, Brian West > > wrote: > > > NO. You want something that people THINK exists and works well... > > Reliable human/voice detection doesn't exist in ANY form. > > I beg to differ. See > http://www.freepatentsonline.com/5521967.html for one way to do > it. It works rather well and can quickly descriminate between > voice and tone. I've no idea who owns that patent now (not me, > for sure). > > There is a simpler, less reliable way of differentiating voice > from tone, that as far as I know isn't patented. If you compare > the RMS power levels of sequential 40 mS periods, call progress > tones will have very consistent power levels from sample to > sample. So if 5 or more 40 mS periods have about the same power > measurement (within say, 2%), it's a tone. Voice will have > dramatic power level differences over that same period. This > works very well in today's telephony environment, where tones are > computer generated. In the old days when ringback tone was > generated off the audio hum from the 20 Hz ring voltage > generator...not so well. > > --Woof! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveu at coppice.org Mon Mar 2 16:32:53 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 03 Mar 2009 08:32:53 +0800 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> Message-ID: <49AC7AB5.5060505@coppice.org> Andy Spitzer wrote: > Woof! > > On Sun, 01 Mar 2009 21:28:18 -0500, Brian West wrote: > > >> NO. You want something that people THINK exists and works well... >> Reliable human/voice detection doesn't exist in ANY form. >> > > I beg to differ. See http://www.freepatentsonline.com/5521967.html for one way to do it. It works rather well and can quickly descriminate between voice and tone. I've no idea who owns that patent now (not me, for sure). > I just had a look through that patent. Its amazing. There is a lot of focussed descriptive text, but a patent only really consists of its claims. Those claims are astonishingly open-ended, and characterise what people had been doing for many years before it was filed - "Well we, er, make a call, we listen for some beeping, and we may hang up based on that". That is a really sick patent. Steve From mszlazak at aol.com Mon Mar 2 16:49:09 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 02 Mar 2009 19:49:09 -0500 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AC6EE4.9080509@gmx.net> References: <49AC6EE4.9080509@gmx.net> Message-ID: <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> I think you need to talk to Brian. Apparently this is a "new" pocketsphinx which works on a different format from those found in the pizza demo. Also, pocketsphinx crashes if it "hears" anything outside the grammar which apparently is a longstanding bug. Brian mentioned they are working on getting this fixed. I kept getting: 2009-02-25 19:49:32 [ERR] mod_pocketsphinx.c:140 pocketsphinx_asr_load_grammar() Can't open dictionary C:\Source\freeswitch-snapshot\Debug\grammar\default.dic. 2009-02-25 19:49:32 [WARNING] mod_pocketsphinx.c:219 pocketsphinx_asr_close() Port Closed. The suggestion was to "Just copy the cmudict.0.6d to default.dic, not sure how well it will perform on windows.. if it does badly you can slim the dictionary down to words you know you'll be using." https://cmusphinx.svn.sourceforge.net/svnroot/cmusphinx/trunk/cmudict/cmudict.0.6d That gave me more problems so I'm waiting for the fix. Mark. -----Original Message----- From: Peter P GMX To: freeswitch-users at lists.freeswitch.org Sent: Mon, 2 Mar 2009 3:42 pm Subject: Re: [Freeswitch-users] pocketsphinx and event socket Thanks Addison. The Pizza files are there (as mentionned is it a copy of an already working system). In fact freeswitch is complaning about /usr/local/freeswitch/grammar/model/communicator which he cannot load So somehow freeswitch is not willing to open the files, but I have no clue why. So any hints are welcome. Best regards Peter Addison Martin schrieb: > Peter, > > You need the grammar files for the pizza demo: > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo > has lonks to premade fles for everyhting to get the pizza demo working > with pocketshinx. Those to not come with the source code when you > update from SVN. > > Nik > > > > On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX wrote: > >> Some more info: >> the system I am working on is a copy (dd copy) of a system where the >> pizza demo works on. >> The only thing I changed was to update to the current freeswitch trunk >> 12293 (it was 10003 before). >> >> Do I need to update the model? I did a make in the model directory, but >> no change. >> >> Best regards >> Peter >> >> Peter P GMX schrieb: >> >>> Hello Brian, >>> >>> thanks for the info. I am a step further, but it cannot load the grammar >>> files. >>> I am sending through event_socket: >>> >>> SendMsg >>> call-command: execute >>> execute-app-name: detect_speech >>> execute-app-arg: pocketsphinx yes no >>> >>> However I get the message (also when I am using Pizza demo): >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() >>> sofia/internal/1000 at sip2.server.com Command Execute >>> detect_speech(pocketsphinx yes no) >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 >>> pocketsphinx_asr_load_grammar() Can't open language model >>> /usr/local/freeswitch/grammar/model/communicator. >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 >>> switch_ivr_detect_speech() Error loading Grammar >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 >>> pocketsphinx_asr_close() Port Closed. >>> >>> However the grammar files are there: >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al >>> total 12752 >>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . >>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. >>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING >>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params >>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means >>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict >>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump >>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances >>> >>> >>> Any hint? >>> >>> Best regards >>> Peter >>> >>> Brian West schrieb: >>> >>> >>>> You can accomplish this .... here is an example using ESL in perl >>>> >>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 >>>> >>>> /b >>>> >>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: >>>> >>>> >>>> >>>> >>>>> Or back to the basics: Is it possible to use pocketsphinx through >>>>> event >>>>> socket? >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090302/27f69544/attachment-0002.html From woof at nortel.com Mon Mar 2 16:51:01 2009 From: woof at nortel.com (Andy Spitzer) Date: Mon, 02 Mar 2009 19:51:01 -0500 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: <49AC7AB5.5060505@coppice.org> References: <49AACD71.5080103@teldio.com> <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org> <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com> <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org> <49AC7AB5.5060505@coppice.org> Message-ID: Woof! On Mon, 02 Mar 2009 19:32:53 -0500, Steve Underwood wrote: I just had a look through that patent. Its amazing. There is a lot of > focussed descriptive text, but a patent only really consists of its > claims. Those claims are astonishingly open-ended, and characterise what > people had been doing for many years before it was filed - "Well we, er, > make a call, we listen for some beeping, and we may hang up based on > that". That is a really sick patent. Yep, I agree. It was the ferping lawyers who kept "adding value" to try to broaden it. What we (the inventors) wrote up was nice and clean. It does have some new and novel technical approaches that we really did come up with...and could find no prior art for. Then the lawyers got to it. A true example of what's wrong with software patents these days. --Woof! From kawarod at laposte.net Mon Mar 2 23:07:32 2009 From: kawarod at laposte.net (rod) Date: Tue, 03 Mar 2009 11:07:32 +0400 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: References: <49AB8562.4050806@laposte.net> Message-ID: <49ACD734.7000700@laposte.net> Hi Michael, I checked on wiki, is the following the good way to go (sorry I'm not very familiar with your debugging tool). $ gdb bin/freeswitch core.xxx bt bt full thread apply all bt thread apply all bt full If I understand well I have to rerun the tests, as I did not start FS using GDB. regards, rod Michael Jerris wrote: > Could you please post this to jira along with a thread apply all bt of > a core file taken from the process with the stuck sessions. > > Mike > > On Mar 2, 2009, at 2:06 AM, rod wrote: > > >> Hi All, >> >> I ran some longer tests with FS 1.0.3 acting as an SBC. >> The test machine has the following specs: >> - Intel Quad Core Q9550 >> - 8GB RAM (far too much from what I saw) >> >> After 3 days running SIPP with 750 simultaneous calls (1500 >> channels) at >> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. >> >> In the CLI, using status command I got this: >> >> freeswitch at internal> status >> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds, >> 607 microseconds >> 15817560 session(s) since startup >> 22 session(s) 0/500 >> >> But when I use "show channels" or "show calls", I see nothing. So I'm >> wondering where are these 22 sessions ? >> >> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs. >> >> Successful call --> 5271434 >> Failed call ---> 1554 (less than 0.03%) >> >> regards, >> rod. >> >> >> >> complete SIPP summary: >> >> ------------------------------ Scenario Screen -------- [1-9]: Change >> Screen -- >> Call-rate(length) Port Total-time Total-calls Remote-host >> 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 >> 10.10.10.254:5060(UDP) >> >> 0 new calls during 0.856 s period 7 ms scheduler resolution >> 0 calls (limit 750) Peak was 750 calls, after 15 s >> 0 Running, 34 Paused, 0 Woken up >> 15544 out-of-call msg (discarded) >> 1 open sockets >> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate >> (kB/s) >> 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate >> (kB/s) >> >> Messages Retrans Timeout >> Unexpected-Msg >> INVITE ----------> 5273022 0 0 >> 100 <---------- 5273022 0 1554 >> 180 <---------- 0 0 0 >> 183 <---------- 0 0 0 >> 200 <---------- E-RTD1 5271434 0 0 >> ACK ----------> 5271434 0 >> Pause [ 35.0s] 5271434 0 >> BYE ----------> 5271434 0 0 >> 200 <---------- 5271434 0 0 >> >> ------------------------------ Test Terminated >> -------------------------------- >> >> >> ----------------------------- Statistics Screen ------- [1-9]: Change >> Screen -- >> Start Time | 2009-02-27 >> 09:11:31 >> Last Reset Time | 2009-03-02 >> 07:49:10 >> Current Time | 2009-03-02 >> 07:49:11 >> -------------------------+--------------------------- >> +-------------------------- >> Counter Name | Periodic value | Cumulative value >> -------------------------+--------------------------- >> +-------------------------- >> Elapsed Time | 00:00:00:857 | >> 70:37:39:429 >> Call Rate | 0.000 cps | 20.739 >> cps >> -------------------------+--------------------------- >> +-------------------------- >> Incoming call created | 0 | >> 0 >> OutGoing call created | 0 | >> 5273022 >> Total Call created | | >> 5273022 >> Current Call | 34 >> | >> -------------------------+--------------------------- >> +-------------------------- >> Successful call | 0 | >> 5271434 >> Failed call | 0 | >> 1554 >> -------------------------+--------------------------- >> +-------------------------- >> Response Time 1 | 00:00:00:000 | >> 00:00:00:240 >> Call Length | 38:32:13:386 | >> 00:00:36:131 >> ------------------------------ Test Terminated >> -------------------------------- >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From mrene_lists at avgs.ca Mon Mar 2 23:56:01 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 3 Mar 2009 02:56:01 -0500 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: <49ACD734.7000700@laposte.net> References: <49AB8562.4050806@laposte.net> <49ACD734.7000700@laposte.net> Message-ID: <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca> Yes, you may also link (or copy) the .gdbinit file found in the support-d folder to your home directory. This is going to enable some GDB macros written for FS. Once thats done you can do the following commands and include them too: list_sessions hash_it_str_x session_manager.session_table switch_core_session_t channel->state Its important to know that what you see in "show channels" and "show calls" is just a DB query to sqlite, Those commands will go directly in the core and list those sessions. Math On 3-Mar-09, at 2:07 AM, rod wrote: > Hi Michael, > > I checked on wiki, is the following the good way to go (sorry I'm not > very familiar with your debugging tool). > > $ gdb bin/freeswitch core.xxx > > bt > bt full > thread apply all bt > thread apply all bt full > > > If I understand well I have to rerun the tests, as I did not start FS > using GDB. > > regards, > rod > > > > > Michael Jerris wrote: >> Could you please post this to jira along with a thread apply all bt >> of >> a core file taken from the process with the stuck sessions. >> >> Mike >> >> On Mar 2, 2009, at 2:06 AM, rod wrote: >> >> >>> Hi All, >>> >>> I ran some longer tests with FS 1.0.3 acting as an SBC. >>> The test machine has the following specs: >>> - Intel Quad Core Q9550 >>> - 8GB RAM (far too much from what I saw) >>> >>> After 3 days running SIPP with 750 simultaneous calls (1500 >>> channels) at >>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. >>> >>> In the CLI, using status command I got this: >>> >>> freeswitch at internal> status >>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 >>> milliseconds, >>> 607 microseconds >>> 15817560 session(s) since startup >>> 22 session(s) 0/500 >>> >>> But when I use "show channels" or "show calls", I see nothing. So >>> I'm >>> wondering where are these 22 sessions ? >>> >>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free >>> CPUs. >>> >>> Successful call --> 5271434 >>> Failed call ---> 1554 (less than 0.03%) >>> >>> regards, >>> rod. >>> >>> >>> >>> complete SIPP summary: >>> >>> ------------------------------ Scenario Screen -------- [1-9]: >>> Change >>> Screen -- >>> Call-rate(length) Port Total-time Total-calls Remote-host >>> 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 >>> 10.10.10.254:5060(UDP) >>> >>> 0 new calls during 0.856 s period 7 ms scheduler resolution >>> 0 calls (limit 750) Peak was 750 calls, after >>> 15 s >>> 0 Running, 34 Paused, 0 Woken up >>> 15544 out-of-call msg (discarded) >>> 1 open sockets >>> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP >>> rate >>> (kB/s) >>> 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate >>> (kB/s) >>> >>> Messages Retrans Timeout >>> Unexpected-Msg >>> INVITE ----------> 5273022 0 0 >>> 100 <---------- 5273022 0 1554 >>> 180 <---------- 0 0 0 >>> 183 <---------- 0 0 0 >>> 200 <---------- E-RTD1 5271434 0 0 >>> ACK ----------> 5271434 0 >>> Pause [ 35.0s] 5271434 0 >>> BYE ----------> 5271434 0 0 >>> 200 <---------- 5271434 0 0 >>> >>> ------------------------------ Test Terminated >>> -------------------------------- >>> >>> >>> ----------------------------- Statistics Screen ------- [1-9]: >>> Change >>> Screen -- >>> Start Time | 2009-02-27 >>> 09:11:31 >>> Last Reset Time | 2009-03-02 >>> 07:49:10 >>> Current Time | 2009-03-02 >>> 07:49:11 >>> -------------------------+--------------------------- >>> +-------------------------- >>> Counter Name | Periodic value | Cumulative >>> value >>> -------------------------+--------------------------- >>> +-------------------------- >>> Elapsed Time | 00:00:00:857 | >>> 70:37:39:429 >>> Call Rate | 0.000 cps | 20.739 >>> cps >>> -------------------------+--------------------------- >>> +-------------------------- >>> Incoming call created | 0 | >>> 0 >>> OutGoing call created | 0 | >>> 5273022 >>> Total Call created | | >>> 5273022 >>> Current Call | 34 >>> | >>> -------------------------+--------------------------- >>> +-------------------------- >>> Successful call | 0 | >>> 5271434 >>> Failed call | 0 | >>> 1554 >>> -------------------------+--------------------------- >>> +-------------------------- >>> Response Time 1 | 00:00:00:000 | >>> 00:00:00:240 >>> Call Length | 38:32:13:386 | >>> 00:00:36:131 >>> ------------------------------ Test Terminated >>> -------------------------------- >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Tue Mar 3 00:16:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 03 Mar 2009 09:16:02 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> Message-ID: <49ACE742.5090809@ewetel.de> Hi Mike, no, I want just one file in stereo, where left channel is caller and right channel is callee. regards Helmut On 02.03.2009 19:05, Michael Collins wrote: > > Just to confirm - you are trying to record each leg of the call into a > separate file? In other words, one call creates two separate audio > recordings? > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090303/d8383e93/attachment-0002.html From Prometheus001 at gmx.net Tue Mar 3 03:07:39 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 03 Mar 2009 12:07:39 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> Message-ID: <49AD0F7B.7000802@gmx.net> Thanks Mark, I now switched back to rev. 10003 and the Pizza works again. Best regards Petere mszlazak at aol.com schrieb: > I think you need to talk to Brian. > > Apparently this is a "new" pocketsphinx which works on a different > format from those found in the pizza demo. > > Also, pocketsphinx crashes if it "hears" anything outside the grammar > which apparently is a longstanding bug. Brian mentioned they are > working on getting this fixed. > > I kept getting: > > 2009-02-25 19:49:32 [ERR] mod_pocketsphinx.c:140 > pocketsphinx_asr_load_grammar() Can't open dictionary > C:\Source\freeswitch-snapshot\Debug\grammar\default.dic. > 2009-02-25 19:49:32 [WARNING] mod_pocketsphinx.c:219 > pocketsphinx_asr_close() Port Closed. > > The suggestion was to "Just copy the cmudict.0.6d to default.dic, not > sure how well it will perform on windows.. if it does badly you can > slim the dictionary down to words you know you'll be using." > > https://cmusphinx.svn.sourceforge.net/svnroot/cmusphinx/trunk/cmudict/cmudict.0.6d > > That gave me more problems so I'm waiting for the fix. > > Mark. > > > > -----Original Message----- > From: Peter P GMX > To: freeswitch-users at lists.freeswitch.org > Sent: Mon, 2 Mar 2009 3:42 pm > Subject: Re: [Freeswitch-users] pocketsphinx and event socket > > Thanks Addison. > > > > > > > > The Pizza files are there (as mentionned is it a copy of an already > > > > > > > > working system). > > > > > > > > In fact freeswitch is complaning about > > > > > > > > /usr/local/freeswitch/grammar/model/communicator which he cannot load > > > > > > > > > > > > > > > > So somehow freeswitch is not willing to open the files, but I have no > > > > > > > > clue why. So any hints are welcome. > > > > > > > > > > > > > > > > Best regards > > > > > > > > Peter > > > > > > > > > > > > > > > > > > > > > > > > Addison Martin schrieb: > > > > > > > > > Peter, > > > > > > > > > > > > > > > > > > You need the grammar files for the pizza demo: > > > > > > > > > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo > > > > > > > > > has lonks to premade fles for everyhting to get the pizza demo working > > > > > > > > > with pocketshinx. Those to not come with the source code when you > > > > > > > > > update from SVN. > > > > > > > > > > > > > > > > > > Nik > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX > wrote: > > > > > > > > > > > > > > > > > >> Some more info: > > > > > > > > >> the system I am working on is a copy (dd copy) of a system where the > > > > > > > > >> pizza demo works on. > > > > > > > > >> The only thing I changed was to update to the current freeswitch trunk > > > > > > > > >> 12293 (it was 10003 before). > > > > > > > > >> > > > > > > > > >> Do I need to update the model? I did a make in the model directory, but > > > > > > > > >> no change. > > > > > > > > >> > > > > > > > > >> Best regards > > > > > > > > >> Peter > > > > > > > > >> > > > > > > > > >> Peter P GMX schrieb: > > > > > > > > >> > > > > > > > > >>> Hello Brian, > > > > > > > > >>> > > > > > > > > >>> thanks for the info. I am a step further, but it cannot load the grammar > > > > > > > > >>> files. > > > > > > > > >>> I am sending through event_socket: > > > > > > > > >>> > > > > > > > > >>> SendMsg > > > > > > > > >>> call-command: execute > > > > > > > > >>> execute-app-name: detect_speech > > > > > > > > >>> execute-app-arg: pocketsphinx yes no > > > > > > > > >>> > > > > > > > > >>> However I get the message (also when I am using Pizza demo): > > > > > > > > >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() > > > > > > > > >>> sofia/internal/1000 at sip2.server.com Command Execute > > > > > > > > >>> detect_speech(pocketsphinx yes no) > > > > > > > > >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 > > > > > > > > >>> pocketsphinx_asr_load_grammar() Can't open language model > > > > > > > > >>> /usr/local/freeswitch/grammar/model/communicator. > > > > > > > > >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 > > > > > > > > >>> switch_ivr_detect_speech() Error loading Grammar > > > > > > > > >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 > > > > > > > > >>> pocketsphinx_asr_close() Port Closed. > > > > > > > > >>> > > > > > > > > >>> However the grammar files are there: > > > > > > > > >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# > > > > > > > > >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al > > > > > > > > >>> total 12752 > > > > > > > > >>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . > > > > > > > > >>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices > > > > > > > > >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances > > > > > > > > >>> > > > > > > > > >>> > > > > > > > > >>> Any hint? > > > > > > > > >>> > > > > > > > > >>> Best regards > > > > > > > > >>> Peter > > > > > > > > >>> > > > > > > > > >>> Brian West schrieb: > > > > > > > > >>> > > > > > > > > >>> > > > > > > > > >>>> You can accomplish this .... here is an example using ESL in perl > > > > > > > > >>>> > > > > > > > > >>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 > > > > > > > > >>>> > > > > > > > > >>>> /b > > > > > > > > >>>> > > > > > > > > >>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: > > > > > > > > >>>> > > > > > > > > >>>> > > > > > > > > >>>> > > > > > > > > >>>> > > > > > > > > >>>>> Or back to the basics: Is it possible to use pocketsphinx through > > > > > > > > >>>>> event > > > > > > > > >>>>> socket? > > > > > > > > >>>>> > > > > > > > > >>>>> > > > > > > > > >>>>> > > > > > > > > >>>> _______________________________________________ > > > > > > > > >>>> Freeswitch-users mailing list > > > > > > > > >>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > http://www.freeswitch.org > > > > > > > > > ------------------------------------------------------------------------ > *A Good Credit Score is 700 or Above. See yours in just 2 easy steps! > * > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Mar 3 06:22:42 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 08:22:42 -0600 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49ACE742.5090809@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> Message-ID: That is exactly what it does. I have confirmed it works can you please file a jira with examples and info of what your experiencing? /b On Mar 3, 2009, at 2:16 AM, Helmut Kuper wrote: > Hi Mike, > > no, I want just one file in stereo, where left channel is caller and > right channel is callee. > > regards > Helmut From brian at freeswitch.org Tue Mar 3 06:23:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 08:23:26 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AD0F7B.7000802@gmx.net> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> Message-ID: <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> It works if you have the latest SVN with the new grammar files in jsgf format. http://www.bkw.org/pizza_gram.tar.gz /b On Mar 3, 2009, at 5:07 AM, Peter P GMX wrote: > Thanks Mark, > > I now switched back to rev. 10003 and the Pizza works again. > > Best regards > Petere From brian at freeswitch.org Tue Mar 3 06:24:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 08:24:07 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49AC6EE4.9080509@gmx.net> References: <49A92BAE.4090907@gmx.net> <49AB0BCD.8030108@gmx.net> <49AC4233.6060506@gmx.net> <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com> <49AC6EE4.9080509@gmx.net> Message-ID: Looks like the acoustical model wasn't installed... you might need to remove all references of pocketsphinx and sphinxbase from libs and let it redownload them all. /b On Mar 2, 2009, at 5:42 PM, Peter P GMX wrote: > Thanks Addison. > The Pizza files are there (as mentionned is it a copy of an already > working system). > In fact freeswitch is complaning about > /usr/local/freeswitch/grammar/model/communicator which he cannot load > > So somehow freeswitch is not willing to open the files, but I have no > clue why. So any hints are welcome. > > Best regards > Peter From helmut.kuper at ewetel.de Tue Mar 3 07:24:26 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 03 Mar 2009 16:24:26 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> Message-ID: <49AD4BAA.8090208@ewetel.de> Hello Brian, you are right. It works, had to use a different player. But caller is also there much louder than callee. Any way to tune this? regards helmut On 03.03.2009 15:22, Brian West wrote: > That is exactly what it does. I have confirmed it works can you > please file a jira with examples and info of what your experiencing? From brian at freeswitch.org Tue Mar 3 07:40:25 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 09:40:25 -0600 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49AD4BAA.8090208@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> Message-ID: <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> Thats going to depend on a lot of issues... are you on SVN trunk? What codecs? What is the path? /b On Mar 3, 2009, at 9:24 AM, Helmut Kuper wrote: > Hello Brian, > > you are right. It works, had to use a different player. But caller is > also there much louder than callee. Any way to tune this? > > regards > helmut From kerrada2003 at yahoo.com Tue Mar 3 07:53:45 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Tue, 3 Mar 2009 07:53:45 -0800 (PST) Subject: [Freeswitch-users] Mandatory IE missing In-Reply-To: Message-ID: <590483.92469.qm@web33703.mail.mud.yahoo.com> Hi, During proxy authentication, I got "mandatory IE missing" error in the response as shown below. How this error can be resolved? recv 585 bytes from udp/[209.82.10.250]:1898 at 15:31:39.933782: ?? ------------------------------------------------------------------------ ?? INVITE sip:9999 at 209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: sip:9999 at 209.82.10.235 ?? Contact: sip:1002 at 209.82.10.250:1091 ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988828 INVITE ?? Content-Length: 187 ?? Content-Type: application/sdp ?? User-Agent: HelpCaster SoftPHONE ?? Supported: com.hearme.mux ? ?? v=0 ?? o=HelpCaster 153553387 153553387 IN IP4 192.168.10.31 ?? s=HelpCaster ?? c=IN IP4 209.82.10.250 ?? t=0 0 ?? m=audio 8002 RTP/AVP 0 4 101 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-15 ?? ------------------------------------------------------------------------ send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.934100: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: sip:9999 at 209.82.10.235 ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988828 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ send 746 bytes to udp/[209.82.10.250]:1091 at 15:31:39.938913: ?? ------------------------------------------------------------------------ ?? SIP/2.0 407 Proxy Authentication Required ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: ;tag=K83mZmZH8g3Hr ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988828 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Accept: application/sdp ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer ?? Proxy-Authenticate: Digest realm="209.82.10.235", nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54" ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.949517: ?? ------------------------------------------------------------------------ ?? ACK sip:9999 at 209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: ;tag=K83mZmZH8g3Hr ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988828 ACK ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 584 bytes from udp/[209.82.10.250]:1898 at 15:31:39.953137: ?? ------------------------------------------------------------------------ ?? INVITE sip:9999 at 209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: sip:9999 at 209.82.10.235 ?? Contact: sip:1002 at 209.82.10.250:1091 ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988829 INVITE ?? Content-Length: 187 ?? Content-Type: application/sdp ?? User-Agent: HelpCaster SoftPHONE ?? Supported: com.hearme.mux ?? Proxy-Authorization:? Digest username="1002",realm="209.82.10.235",nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54",response=" d1614cd024acab0b794751285f5cb1fe",uri="sip:9999 at 209.82.10.235" ? ?? ------------------------------------------------------------------------ send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.953336: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: sip:9999 at 209.82.10.235 ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988829 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ send 696 bytes to udp/[209.82.10.250]:1091 at 15:31:39.956016: ?? ------------------------------------------------------------------------ ?? SIP/2.0 480 Temporarily Unavailable ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: ;tag=mHXD1FgN5SS4K ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988829 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Accept: application/sdp ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer ?? Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING" ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.964632: ?? ------------------------------------------------------------------------ ?? ACK sip:9999 at 209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1091 ?? From: HC-Desktop ;tag=55409708 ?? To: ;tag=mHXD1FgN5SS4K ?? Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 ?? CSeq: 421988829 ACK ?? Content-Length: 0 Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090303/39844287/attachment-0002.html From freeswitch at servercorps.com Tue Mar 3 08:42:01 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Tue, 3 Mar 2009 10:42:01 -0600 Subject: [Freeswitch-users] First time test set up FreeSwitch and SPA3102/SPA3000 In-Reply-To: <9ed22e920903012343u5e50c89fkfa9d1127894199c@mail.gmail.com> References: <9ed22e920903012343u5e50c89fkfa9d1127894199c@mail.gmail.com> Message-ID: <92e7d2090903030842p509573co969f89a3fb72a8ad@mail.gmail.com> api, It may help if we knew a little about what you are trying to do. Could you explain a bit why you think you need a PBX? What re your goals, and objective? Knowing that may help us get you the information you need. Regards, Nik On Mon, Mar 2, 2009 at 1:43 AM, Aplayful Idiot wrote: > I have no background in telephony but probably need to use a PBX. > > FreeSwitch was recommended by a casual contact so I would like to start > first by setting up a small test. > > I have a SPA3102 attached to the box running FS and to a ordinary phone > line. I registered SPA in conf/directory/default/line1.xml and it works ok > but I can't get caller id numbers from incoming calls. All FS sees is > "line1" which is found in file line1.xml as name="effective_caller_id_number" value="line1"/>. > > Looking back over the FS wiki, I'm now wondering if the SPA should of been > set-up as a gateway but reading it is confusing at least to me. Sometimes I > think the analogue-line-SPA-FS is like a softphone which is registered to an > extension numbered xml file in conf/directory/default/ but then issues like > not getting outside incoming caller id's makes me think I've got this all > wrong. > > Can someone help me out with this? > > api > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Tue Mar 3 13:17:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Mar 2009 15:17:30 -0600 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca> References: <49AB8562.4050806@laposte.net> <49ACD734.7000700@laposte.net> <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca> Message-ID: <191c3a030903031317y55ce2c9bsb4a93518db261973@mail.gmail.com> and you may want to update and try trunk to make sure it's not fixed On Tue, Mar 3, 2009 at 1:56 AM, Mathieu Rene wrote: > Yes, you may also link (or copy) the .gdbinit file found in the > support-d folder to your home directory. > This is going to enable some GDB macros written for FS. > > Once thats done you can do the following commands and include them too: > > list_sessions > > hash_it_str_x session_manager.session_table switch_core_session_t > channel->state > > > Its important to know that what you see in "show channels" and "show > calls" is just a DB query to sqlite, Those commands will go directly > in the core and list those sessions. > > Math > > On 3-Mar-09, at 2:07 AM, rod wrote: > > > Hi Michael, > > > > I checked on wiki, is the following the good way to go (sorry I'm not > > very familiar with your debugging tool). > > > > $ gdb bin/freeswitch core.xxx > > > > bt > > bt full > > thread apply all bt > > thread apply all bt full > > > > > > If I understand well I have to rerun the tests, as I did not start FS > > using GDB. > > > > regards, > > rod > > > > > > > > > > Michael Jerris wrote: > >> Could you please post this to jira along with a thread apply all bt > >> of > >> a core file taken from the process with the stuck sessions. > >> > >> Mike > >> > >> On Mar 2, 2009, at 2:06 AM, rod wrote: > >> > >> > >>> Hi All, > >>> > >>> I ran some longer tests with FS 1.0.3 acting as an SBC. > >>> The test machine has the following specs: > >>> - Intel Quad Core Q9550 > >>> - 8GB RAM (far too much from what I saw) > >>> > >>> After 3 days running SIPP with 750 simultaneous calls (1500 > >>> channels) at > >>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. > >>> > >>> In the CLI, using status command I got this: > >>> > >>> freeswitch at internal> status > >>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 > >>> milliseconds, > >>> 607 microseconds > >>> 15817560 session(s) since startup > >>> 22 session(s) 0/500 > >>> > >>> But when I use "show channels" or "show calls", I see nothing. So > >>> I'm > >>> wondering where are these 22 sessions ? > >>> > >>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free > >>> CPUs. > >>> > >>> Successful call --> 5271434 > >>> Failed call ---> 1554 (less than 0.03%) > >>> > >>> regards, > >>> rod. > >>> > >>> > >>> > >>> complete SIPP summary: > >>> > >>> ------------------------------ Scenario Screen -------- [1-9]: > >>> Change > >>> Screen -- > >>> Call-rate(length) Port Total-time Total-calls Remote-host > >>> 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 > >>> 10.10.10.254:5060(UDP) > >>> > >>> 0 new calls during 0.856 s period 7 ms scheduler resolution > >>> 0 calls (limit 750) Peak was 750 calls, after > >>> 15 s > >>> 0 Running, 34 Paused, 0 Woken up > >>> 15544 out-of-call msg (discarded) > >>> 1 open sockets > >>> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP > >>> rate > >>> (kB/s) > >>> 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate > >>> (kB/s) > >>> > >>> Messages Retrans Timeout > >>> Unexpected-Msg > >>> INVITE ----------> 5273022 0 0 > >>> 100 <---------- 5273022 0 1554 > >>> 180 <---------- 0 0 0 > >>> 183 <---------- 0 0 0 > >>> 200 <---------- E-RTD1 5271434 0 0 > >>> ACK ----------> 5271434 0 > >>> Pause [ 35.0s] 5271434 0 > >>> BYE ----------> 5271434 0 0 > >>> 200 <---------- 5271434 0 0 > >>> > >>> ------------------------------ Test Terminated > >>> -------------------------------- > >>> > >>> > >>> ----------------------------- Statistics Screen ------- [1-9]: > >>> Change > >>> Screen -- > >>> Start Time | 2009-02-27 > >>> 09:11:31 > >>> Last Reset Time | 2009-03-02 > >>> 07:49:10 > >>> Current Time | 2009-03-02 > >>> 07:49:11 > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Counter Name | Periodic value | Cumulative > >>> value > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Elapsed Time | 00:00:00:857 | > >>> 70:37:39:429 > >>> Call Rate | 0.000 cps | 20.739 > >>> cps > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Incoming call created | 0 | > >>> 0 > >>> OutGoing call created | 0 | > >>> 5273022 > >>> Total Call created | | > >>> 5273022 > >>> Current Call | 34 > >>> | > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Successful call | 0 | > >>> 5271434 > >>> Failed call | 0 | > >>> 1554 > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Response Time 1 | 00:00:00:000 | > >>> 00:00:00:240 > >>> Call Length | 38:32:13:386 | > >>> 00:00:36:131 > >>> ------------------------------ Test Terminated > >>> -------------------------------- > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090303/bf5f72d7/attachment-0002.html From mike at jerris.com Tue Mar 3 14:08:11 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 3 Mar 2009 17:08:11 -0500 Subject: [Freeswitch-users] Mandatory IE missing In-Reply-To: <590483.92469.qm@web33703.mail.mud.yahoo.com> References: <590483.92469.qm@web33703.mail.mud.yahoo.com> Message-ID: The debug logs should give you more information about what is happening here. Mike On Mar 3, 2009, at 10:53 AM, Ali Al-Rubaie wrote: > Hi, > > During proxy authentication, I got "mandatory IE missing" error in > the response as shown below. How this error can be resolved? From Prometheus001 at gmx.net Tue Mar 3 17:05:40 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 04 Mar 2009 02:05:40 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> Message-ID: <49ADD3E4.20408@gmx.net> Thank you Brian, I will try this later. Currently I was happy to get this working on SVN 10003. As mod_pockesphinx has changed/evolved significantely: Will there also be major changes in the events I receive through mod_eventsocket? I spend some time on parsing the right data out of the eventsocket interface, and I would just have an idea, if I will have to expect significant work to do, when I later switch to the current SVN. Will I need updated grammar files for the other models too? Best regards Peter Brian West schrieb: > It works if you have the latest SVN with the new grammar files in jsgf > format. http://www.bkw.org/pizza_gram.tar.gz > > > /b > > On Mar 3, 2009, at 5:07 AM, Peter P GMX wrote: > > >> Thanks Mark, >> >> I now switched back to rev. 10003 and the Pizza works again. >> >> Best regards >> Petere >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Tue Mar 3 19:00:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 21:00:35 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49ADD3E4.20408@gmx.net> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> <49ADD3E4.20408@gmx.net> Message-ID: <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> Well you should use ESL then ;) /b On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote: > Thank you Brian, > > I will try this later. > > Currently I was happy to get this working on SVN 10003. > > As mod_pockesphinx has changed/evolved significantely: Will there also > be major changes in the events I receive through mod_eventsocket? > I spend some time on parsing the right data out of the eventsocket > interface, and I would just have an idea, if I will have to expect > significant work to do, when I later switch to the current SVN. > > Will I need updated grammar files for the other models too? > > Best regards > Peter From jgarland at jasongarland.com Tue Mar 3 20:12:45 2009 From: jgarland at jasongarland.com (Jason Garland) Date: Tue, 3 Mar 2009 23:12:45 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Qt portaudio interface In-Reply-To: <3A65F8B6-96B4-45EE-80E9-A30AFEED0BB3@avgs.ca> References: <3A65F8B6-96B4-45EE-80E9-A30AFEED0BB3@avgs.ca> Message-ID: I still think a web based interface would work well and be more cross platform. The web app could be served from within FS. http://127.0.0.1:8080/myfancysoftphone/ Sent from my iPhone On Mar 2, 2009, at 12:00 AM, Mathieu Rene wrote: > Hi all, > > Anyone interested in contributing to a Qt interface in order to make a > decent softphone using FS please reply to this thread. > (also give your availability so we can have a conference call to > decide stuff) > > Thanks, > Math > > _______________________________________________ > 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 jgarland at jasongarland.com Tue Mar 3 20:18:40 2009 From: jgarland at jasongarland.com (Jason Garland) Date: Tue, 3 Mar 2009 23:18:40 -0500 Subject: [Freeswitch-users] Mandatory IE missing In-Reply-To: <590483.92469.qm@web33703.mail.mud.yahoo.com> References: <590483.92469.qm@web33703.mail.mud.yahoo.com> Message-ID: <0744A0FC-7FCE-4662-944D-9FE133142AE1@jasongarland.com> Your HelpCaster softphone didn't send any SDP on the second INVITE. Fix the softphone or try turning on late codec negotiation in you Sofia profile. Sent from my iPhone On Mar 3, 2009, at 10:53 AM, Ali Al-Rubaie wrote: > > Hi, > > During proxy authentication, I got "mandatory IE missing" error in > the response as shown below. How this error can be resolved? > > recv 585 bytes from udp/[209.82.10.250]:1898 at 15:31:39.933782: > > > --- > --------------------------------------------------------------------- > > INVITE sip:9999 at 209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: sip:9999 at 209.82.10.235 > > Contact: sip:1002 at 209.82.10.250:1091 > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988828 INVITE > > Content-Length: 187 > > Content-Type: application/sdp > > User-Agent: HelpCaster SoftPHONE > > Supported: com.hearme.mux > > > > v=0 > > o=HelpCaster 153553387 153553387 IN IP4 192.168.10.31 > > s=HelpCaster > > c=IN IP4 209.82.10.250 > > t=0 0 > > m=audio 8002 RTP/AVP 0 4 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > --- > --------------------------------------------------------------------- > > send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.934100: > > > --- > --------------------------------------------------------------------- > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: sip:9999 at 209.82.10.235 > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988828 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > send 746 bytes to udp/[209.82.10.250]:1091 at 15:31:39.938913: > > > --- > --------------------------------------------------------------------- > > SIP/2.0 407 Proxy Authentication Required > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: ;tag=K83mZmZH8g3Hr > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988828 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > > Proxy-Authenticate: Digest realm="209.82.10.235", > nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54" > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.949517: > > > --- > --------------------------------------------------------------------- > > ACK sip:9999 at 209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: ;tag=K83mZmZH8g3Hr > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988828 ACK > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > recv 584 bytes from udp/[209.82.10.250]:1898 at 15:31:39.953137: > > > --- > --------------------------------------------------------------------- > > INVITE sip:9999 at 209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: sip:9999 at 209.82.10.235 > > Contact: sip:1002 at 209.82.10.250:1091 > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988829 INVITE > > Content-Length: 187 > > Content-Type: application/sdp > > User-Agent: HelpCaster SoftPHONE > > Supported: com.hearme.mux > > Proxy-Authorization: Digest > username= > "1002" > ,realm= > "209.82.10.235" > ,nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54",response=" > > d1614cd024acab0b794751285f5cb1fe",uri="sip:9999 at 209.82.10.235" > > > > > --- > --------------------------------------------------------------------- > > send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.953336: > > > --- > --------------------------------------------------------------------- > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: sip:9999 at 209.82.10.235 > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988829 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > send 696 bytes to udp/[209.82.10.250]:1091 at 15:31:39.956016: > > > --- > --------------------------------------------------------------------- > > SIP/2.0 480 Temporarily Unavailable > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: ;tag=mHXD1FgN5SS4K > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988829 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > > Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING" > > Content-Length: 0 > > > > > --- > --------------------------------------------------------------------- > > recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.964632: > > > --- > --------------------------------------------------------------------- > > ACK sip:9999 at 209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1091 > > From: HC-Desktop ;tag=55409708 > > To: ;tag=mHXD1FgN5SS4K > > Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31 > > CSeq: 421988829 ACK > > Content-Length: 0 > > > Thanks! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090303/3d1cc9e8/attachment-0002.html From brian at freeswitch.org Tue Mar 3 20:21:08 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Mar 2009 22:21:08 -0600 Subject: [Freeswitch-users] Mandatory IE missing In-Reply-To: <0744A0FC-7FCE-4662-944D-9FE133142AE1@jasongarland.com> References: <590483.92469.qm@web33703.mail.mud.yahoo.com> <0744A0FC-7FCE-4662-944D-9FE133142AE1@jasongarland.com> Message-ID: <3E391AC2-BF15-4987-B143-5CF05855693E@freeswitch.org> Jason thanks.. you know I think we all missed that one. /b On Mar 3, 2009, at 10:18 PM, Jason Garland wrote: > Your HelpCaster softphone didn't send any SDP on the second INVITE. > Fix the softphone or try turning on late codec negotiation in you > Sofia profile. > > Sent from my iPhone From kawarod at laposte.net Tue Mar 3 22:47:21 2009 From: kawarod at laposte.net (rod) Date: Wed, 04 Mar 2009 10:47:21 +0400 Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test In-Reply-To: <191c3a030903031317y55ce2c9bsb4a93518db261973@mail.gmail.com> References: <49AB8562.4050806@laposte.net> <49ACD734.7000700@laposte.net> <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca> <191c3a030903031317y55ce2c9bsb4a93518db261973@mail.gmail.com> Message-ID: <49AE23F9.7050108@laposte.net> Hi, I'm already trying this. I will update this thread next week. regards. Anthony Minessale wrote: > and you may want to update and try trunk to make sure it's not fixed > > On Tue, Mar 3, 2009 at 1:56 AM, Mathieu Rene > wrote: > > Yes, you may also link (or copy) the .gdbinit file found in the > support-d folder to your home directory. > This is going to enable some GDB macros written for FS. > > Once thats done you can do the following commands and include them > too: > > list_sessions > > hash_it_str_x session_manager.session_table switch_core_session_t > channel->state > > > Its important to know that what you see in "show channels" and "show > calls" is just a DB query to sqlite, Those commands will go directly > in the core and list those sessions. > > Math > > On 3-Mar-09, at 2:07 AM, rod wrote: > > > Hi Michael, > > > > I checked on wiki, is the following the good way to go (sorry > I'm not > > very familiar with your debugging tool). > > > > $ gdb bin/freeswitch core.xxx > > > > bt > > bt full > > thread apply all bt > > thread apply all bt full > > > > > > If I understand well I have to rerun the tests, as I did not > start FS > > using GDB. > > > > regards, > > rod > > > > > > > > > > Michael Jerris wrote: > >> Could you please post this to jira along with a thread apply all bt > >> of > >> a core file taken from the process with the stuck sessions. > >> > >> Mike > >> > >> On Mar 2, 2009, at 2:06 AM, rod wrote: > >> > >> > >>> Hi All, > >>> > >>> I ran some longer tests with FS 1.0.3 acting as an SBC. > >>> The test machine has the following specs: > >>> - Intel Quad Core Q9550 > >>> - 8GB RAM (far too much from what I saw) > >>> > >>> After 3 days running SIPP with 750 simultaneous calls (1500 > >>> channels) at > >>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP. > >>> > >>> In the CLI, using status command I got this: > >>> > >>> freeswitch at internal> status > >>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 > >>> milliseconds, > >>> 607 microseconds > >>> 15817560 session(s) since startup > >>> 22 session(s) 0/500 > >>> > >>> But when I use "show channels" or "show calls", I see nothing. So > >>> I'm > >>> wondering where are these 22 sessions ? > >>> > >>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free > >>> CPUs. > >>> > >>> Successful call --> 5271434 > >>> Failed call ---> 1554 (less than 0.03%) > >>> > >>> regards, > >>> rod. > >>> > >>> > >>> > >>> complete SIPP summary: > >>> > >>> ------------------------------ Scenario Screen -------- [1-9]: > >>> Change > >>> Screen -- > >>> Call-rate(length) Port Total-time Total-calls Remote-host > >>> 50.0(35000 ms)/1.000s 5060 254259.42 s 5273022 > >>> 10.10.10.254:5060(UDP) > >>> > >>> 0 new calls during 0.856 s period 7 ms scheduler resolution > >>> 0 calls (limit 750) Peak was 750 calls, after > >>> 15 s > >>> 0 Running, 34 Paused, 0 Woken up > >>> 15544 out-of-call msg (discarded) > >>> 1 open sockets > >>> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP > >>> rate > >>> (kB/s) > >>> 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate > >>> (kB/s) > >>> > >>> Messages Retrans Timeout > >>> Unexpected-Msg > >>> INVITE ----------> 5273022 0 0 > >>> 100 <---------- 5273022 0 1554 > >>> 180 <---------- 0 0 0 > >>> 183 <---------- 0 0 0 > >>> 200 <---------- E-RTD1 5271434 0 0 > >>> ACK ----------> 5271434 0 > >>> Pause [ 35.0s] 5271434 0 > >>> BYE ----------> 5271434 0 0 > >>> 200 <---------- 5271434 0 0 > >>> > >>> ------------------------------ Test Terminated > >>> -------------------------------- > >>> > >>> > >>> ----------------------------- Statistics Screen ------- [1-9]: > >>> Change > >>> Screen -- > >>> Start Time | 2009-02-27 > >>> 09:11:31 > >>> Last Reset Time | 2009-03-02 > >>> 07:49:10 > >>> Current Time | 2009-03-02 > >>> 07:49:11 > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Counter Name | Periodic value | Cumulative > >>> value > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Elapsed Time | 00:00:00:857 | > >>> 70:37:39:429 > >>> Call Rate | 0.000 cps | 20.739 > >>> cps > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Incoming call created | 0 | > >>> 0 > >>> OutGoing call created | 0 | > >>> 5273022 > >>> Total Call created | | > >>> 5273022 > >>> Current Call | 34 > >>> | > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Successful call | 0 | > >>> 5271434 > >>> Failed call | 0 | > >>> 1554 > >>> -------------------------+--------------------------- > >>> +-------------------------- > >>> Response Time 1 | 00:00:00:000 | > >>> 00:00:00:240 > >>> Call Length | 38:32:13:386 | > >>> 00:00:36:131 > >>> ------------------------------ Test Terminated > >>> -------------------------------- > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mszlazak at aol.com Tue Mar 3 23:24:10 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 04 Mar 2009 02:24:10 -0500 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net><57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org><49ADD3E4.20408@gmx.net> <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> Message-ID: <8CB6AB158D38B9A-A74-24FC@WEBMAIL-MA07.sysops.aol.com> Brian, Peter says: "mod_pockesphinx has changed/evolved significantely" Since this seems to be coming without any warning, what specifically are all these and future changes and why are they happening? Mark. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tue, 3 Mar 2009 7:00 pm Subject: Re: [Freeswitch-users] pocketsphinx and event socket Well you should use ESL then ;) /b On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote: > Thank you Brian, > > I will try this later. > > Currently I was happy to get this working on SVN 10003. > > As : Will there also > be major changes in the events I receive through mod_eventsocket? > I spend some time on parsing the right data out of the eventsocket > interface, and I would just have an idea, if I will have to expect > significant work to do, when I later switch to the current SVN. > > Will I need updated grammar files for the other models too? > > Best regards > Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/57128233/attachment-0002.html From Claudio.Cavalera at italtel.it Wed Mar 4 03:41:53 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Mar 2009 12:41:53 +0100 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > freeswitch-users-bounces at lists.freeswitch.org wrote: >> Cavalera Claudio Luigi wrote: >>> So could you please someone more expert with me clarify the >>> difference between -d and -l options so that I can update the wiki >>> which is now wrong/incomplete ? >>> >>> -l, --loglevel=command Log Level >>> -q, --quiet Disable logging >>> -d, --debug=level Debug Level (0 - 7) >> >> The difference is that -d controls the level of debugging output >> generated by fs_cli itself. The log level controls which log messages >> from your running FreeSWITCH daemon are printed to the fs_cli >> console. Could you please explain why if I issue this command in fs_cli: "fsctl loglevel warning" to lower the log level in output file freeswitch.log but then I can't have level debug in fs_cli even if I re-set it with "/log debug" The fsctl loglevel (which I guess is the same as the global loglevel in switch.conf.xml) must always be higher than loglevel in fs_cli? If correct, is this true also for the console_loglevel ? Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From helmut.kuper at ewetel.de Wed Mar 4 04:28:52 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 04 Mar 2009 13:28:52 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> Message-ID: <49AE7404.8000905@ewetel.de> Hi, well, both extensions are direct connected to FS, so we have an internal call. Caller is snom 370, callee is snom 820, codecs are G.722 with RTP crypt AES 32, both phones are within the same LAN. FS is in a separate LAN. Energy levels for speakers and micros on both snom phones are the same. I use latest SVN trunk. regards Helmut On 03.03.2009 16:40, Brian West wrote: > Thats going to depend on a lot of issues... are you on SVN trunk? > What codecs? What is the path? > > /b > From mrene_lists at avgs.ca Wed Mar 4 04:49:06 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 07:49:06 -0500 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: References: Message-ID: <0321F1A5-F307-465E-8204-FE9FF4842900@avgs.ca> fsctl loglevel [xxx] tells the core to ignore log messages not at least that level. They wont get logged, you wont see them on the console and no events will be generated. console loglevel [xxx] (on the "real" console) tells mod_console to do that filtering. /log [xxx] tells fs_cli to do the filtering. It all depends where you filter it, if you want to change the logfile's level without affecting anything else, I recommand you edit logfile.conf.xml Math On 4-Mar-09, at 6:41 AM, Cavalera Claudio Luigi wrote: > freeswitch-users-bounces at lists.freeswitch.org wrote: >> freeswitch-users-bounces at lists.freeswitch.org wrote: >>> Cavalera Claudio Luigi wrote: >>>> So could you please someone more expert with me clarify the >>>> difference between -d and -l options so that I can update the wiki >>>> which is now wrong/incomplete ? >>>> >>>> -l, --loglevel=command Log Level >>>> -q, --quiet Disable logging >>>> -d, --debug=level Debug Level (0 - 7) >>> >>> The difference is that -d controls the level of debugging output >>> generated by fs_cli itself. The log level controls which log >>> messages >>> from your running FreeSWITCH daemon are printed to the fs_cli >>> console. > > > Could you please explain why if I issue this command in fs_cli: > "fsctl loglevel warning" > to lower the log level in output file freeswitch.log > but then I can't have level debug in fs_cli even if I re-set it with > "/log debug" > > The fsctl loglevel (which I guess is the same as the global loglevel > in > switch.conf.xml) must always be higher than loglevel in fs_cli? > If correct, is this true also for the console_loglevel ? > > Thanks, > Claudio > > > 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-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Wed Mar 4 05:08:19 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 04 Mar 2009 14:08:19 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49AE7404.8000905@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> Message-ID: <49AE7D43.9070409@ewetel.de> Hello, I just did a further test to nail down the volume problem with recording a call. Caller as well as callee recorded the session in stereo. I got two files in "recordings". In both files, the party which starts the recording seems to be louder and more clear as the opposite party. It seems to me, that the party which didn't start the recording was transcoded to a different codec (guess g711). Remember: Both partys are using g722 which was confirmed by the phone's display. regards helmut On 04.03.2009 13:28, Helmut Kuper wrote: > Hi, > > well, both extensions are direct connected to FS, so we have an internal > call. Caller is snom 370, callee is snom 820, codecs are G.722 with RTP > crypt AES 32, both phones are within the same LAN. FS is in a separate > LAN. Energy levels for speakers and micros on both snom phones are the same. > > I use latest SVN trunk. > > regards > Helmut From Claudio.Cavalera at italtel.it Wed Mar 4 07:03:46 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Mar 2009 16:03:46 +0100 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: <0321F1A5-F307-465E-8204-FE9FF4842900@avgs.ca> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > fsctl loglevel [xxx] tells the core to ignore log messages not at > least that level. They wont get logged, you wont see them on the > console and no events will be generated. > > console loglevel [xxx] (on the "real" console) tells > mod_console to do > that filtering. > /log [xxx] tells fs_cli to do the filtering. > > It all depends where you filter it, if you want to change the > logfile's level without affecting anything else, I recommand > you edit > logfile.conf.xml > > Math Ah ok, so it's better to keep switch.conf.xml to debug and then to adjust the single sources to lower logging levels. I have also put these lines in a rc.freeswitch script before starting freeswitch export SOFIA_DEBUG=9 export NUA_DEBUG=9 export NTA_DEBUG=9 export TPORT_DEBUG=9 export TPORT_LOG=1 becase I wanted to have all the sofia output redirected to a different logfile but it does not work. Instead if on any console I use fs_cli I see there the sofia stuff, how does that work? Thanks, Claudio Internet E. 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From brian at freeswitch.org Wed Mar 4 07:13:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Mar 2009 09:13:01 -0600 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49AE7D43.9070409@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> <49AE7D43.9070409@ewetel.de> Message-ID: <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> Can you email me a sample recording please? /b On Mar 4, 2009, at 7:08 AM, Helmut Kuper wrote: > Hello, > > I just did a further test to nail down the volume problem with > recording > a call. Caller as well as callee recorded the session in stereo. I got > two files in "recordings". In both files, the party which starts the > recording seems to be louder and more clear as the opposite party. It > seems to me, that the party which didn't start the recording was > transcoded to a different codec (guess g711). Remember: Both partys > are > using g722 which was confirmed by the phone's display. > > > regards > helmut From helmut.kuper at ewetel.de Wed Mar 4 07:37:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 04 Mar 2009 16:37:38 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> <49AE7D43.9070409@ewetel.de> <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> Message-ID: <49AEA042.3090908@ewetel.de> Hi Brian, both files are on their way ... Quite big (18MB) regards Helmut On 04.03.2009 16:13, Brian West wrote: > Can you email me a sample recording please? > > /b > > On Mar 4, 2009, at 7:08 AM, Helmut Kuper wrote: > > >> Hello, >> >> I just did a further test to nail down the volume problem with >> recording >> a call. Caller as well as callee recorded the session in stereo. I got >> two files in "recordings". In both files, the party which starts the >> recording seems to be louder and more clear as the opposite party. It >> seems to me, that the party which didn't start the recording was >> transcoded to a different codec (guess g711). Remember: Both partys >> are >> using g722 which was confirmed by the phone's display. >> >> >> regards >> helmut >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/1f04e99b/attachment-0002.html From brian at freeswitch.org Wed Mar 4 07:43:51 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Mar 2009 09:43:51 -0600 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: <49AEA042.3090908@ewetel.de> References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> <49AE7D43.9070409@ewetel.de> <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> <49AEA042.3090908@ewetel.de> Message-ID: Already received them.. Opened them in Audacity.. compared them... listened to them a couple of times... and they sound fine to me.... So answer the two questions I asked in the private email and we'll see ;) /b On Mar 4, 2009, at 9:37 AM, Helmut Kuper wrote: > Hi Brian, > > both files are on their way ... Quite big (18MB) > > regards > Helmut From helmut.kuper at ewetel.de Wed Mar 4 08:54:14 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 04 Mar 2009 17:54:14 +0100 Subject: [Freeswitch-users] How to send CUSTOM event via event socket? Message-ID: <49AEB236.10201@ewetel.de> Hello, how to send a CUSTOM event via event socket? Currently I send this: sendevent CUSTOM\n Event-Name: CUSTOM\n Event-Subclass: myevent::snom-aurl\n Snom-Event: OFFHOOK\n \n I got an +OK reply from FS. I subscribed to ALL event and get only this: Content-Length: 516 Content-Type: text/event-plain Event-Name: COMMAND Core-UUID: 7b2b75a4-08ca-11de-b1f3-3d0cddd2708d FreeSWITCH-Hostname: ippbx-prod-node0 FreeSWITCH-IPv4: 85.16.246.5 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-03-04%2017%3A42%3A35 Event-Date-GMT: Wed,%2004%20Mar%202009%2016%3A42%3A35%20GMT Event-Date-Timestamp: 1236184955199959 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1045 Command: sendevent%20CUSTOM Event-Name: CUSTOM Event-Subclass: ewetel%3A%3Asnom-aurl Snom-Event: OFFHOOK But there is no CUSTOM event received. regards helmut From fax at virgintechnologies.com Wed Mar 4 10:45:45 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Wed, 04 Mar 2009 18:45:45 +0000 Subject: [Freeswitch-users] separate gateways SIP and RTP stream Message-ID: I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060. Thank you Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/c20f8a51/attachment-0002.html From mrene_lists at avgs.ca Wed Mar 4 10:47:47 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 13:47:47 -0500 Subject: [Freeswitch-users] separate gateways SIP and RTP stream In-Reply-To: References: Message-ID: <1AE3D5A9-6753-4005-947A-A92B16F55DF9@avgs.ca> Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. Math On 4-Mar-09, at 1:45 PM, Justin Miller wrote: > I'm setting up a test SIP trunk with Allstream. They are using > separate gateway IPs for SIP signalling and RTP streams. Has anyone > done this with Freeswitch? I was planning on setting them up as an > extension in the internal profile since they use SIP port 5060. > Thank you > Justin > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/7182b36e/attachment-0002.html From fax at virgintechnologies.com Wed Mar 4 11:25:03 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Wed, 04 Mar 2009 19:25:03 +0000 Subject: [Freeswitch-users] separate gateways SIP and RTP stream Message-ID: So is there an "RTP_proxy" parameter that can be set for the gateway, or in my case, the user extension? How would I define the separate IP for the RTP stream. -----Original Message----- From: Mathieu Rene [mailto:mrene_lists at avgs.ca] Sent: Wednesday, March 4, 2009 11:47 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. Math On 4-Mar-09, at 1:45 PM, Justin Miller wrote: I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060. Thank you Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/709abda0/attachment-0002.html From mrene_lists at avgs.ca Wed Mar 4 11:26:10 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 14:26:10 -0500 Subject: [Freeswitch-users] separate gateways SIP and RTP stream In-Reply-To: References: Message-ID: <259504C8-C9CA-4A0E-98E6-DC091734D356@avgs.ca> You don't. The SIP protocol already takes care of that, just make sure you open your firewall properly. On 4-Mar-09, at 2:25 PM, Justin Miller wrote: > So is there an "RTP_proxy" parameter that can be set for the > gateway, or in my case, the user extension? How would I define the > separate IP for the RTP stream. > -----Original Message----- > From: Mathieu Rene [mailto:mrene_lists at avgs.ca] > Sent: Wednesday, March 4, 2009 11:47 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Using different IPs for signalling and media is perfectly legal, > thats why the SDP contains information on where the media should be > sent.. > > > Math > > On 4-Mar-09, at 1:45 PM, Justin Miller wrote: > >> I'm setting up a test SIP trunk with Allstream. They are using >> separate gateway IPs for SIP signalling and RTP streams. Has >> anyone done this with Freeswitch? I was planning on setting them >> up as an extension in the internal profile since they use SIP port >> 5060. >> Thank you >> Justin >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/064d6a11/attachment-0002.html From mrene_lists at avgs.ca Wed Mar 4 11:27:43 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 14:27:43 -0500 Subject: [Freeswitch-users] How to send CUSTOM event via event socket? In-Reply-To: <49AEB236.10201@ewetel.de> References: <49AEB236.10201@ewetel.de> Message-ID: <0C5EA2F5-9B6D-4D00-98D1-2DD60C8E8F59@avgs.ca> On 4-Mar-09, at 11:54 AM, Helmut Kuper wrote: > Event-Name: CUSTOM > Event-Subclass: ewetel%3A%3Asnom-aurl > Snom-Event: OFFHOOK ^^ mod_event_socket loads up commands it receives in an event struct, when you do a "sendevent", it changes the eventid to be changed to the one you specified, unfortunately it doesnt remove the first Event-Name header. Open a JIRA if thats what you want. Math > Hello, > > how to send a CUSTOM event via event socket? Currently I send this: > > sendevent CUSTOM\n > Event-Name: CUSTOM\n > Event-Subclass: myevent::snom-aurl\n > Snom-Event: OFFHOOK\n > \n > > > I got an +OK reply from FS. I subscribed to ALL event and get only > this: > > Content-Length: 516 > Content-Type: text/event-plain > > Event-Name: COMMAND > Core-UUID: 7b2b75a4-08ca-11de-b1f3-3d0cddd2708d > FreeSWITCH-Hostname: ippbx-prod-node0 > FreeSWITCH-IPv4: 85.16.246.5 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-03-04%2017%3A42%3A35 > Event-Date-GMT: Wed,%2004%20Mar%202009%2016%3A42%3A35%20GMT > Event-Date-Timestamp: 1236184955199959 > Event-Calling-File: mod_event_socket.c > Event-Calling-Function: read_packet > Event-Calling-Line-Number: 1045 > Command: sendevent%20CUSTOM > Event-Name: CUSTOM > Event-Subclass: ewetel%3A%3Asnom-aurl > Snom-Event: OFFHOOK > > But there is no CUSTOM event received. > > regards > helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fax at virgintechnologies.com Wed Mar 4 12:25:16 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Wed, 04 Mar 2009 20:25:16 +0000 Subject: [Freeswitch-users] separate gateways SIP and RTP stream Message-ID: Ok, got it. Thank you. -----Original Message----- From: Mathieu Rene [mailto:mrene_lists at avgs.ca] Sent: Wednesday, March 4, 2009 12:26 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream You don't. The SIP protocol already takes care of that, just make sure you open your firewall properly. On 4-Mar-09, at 2:25 PM, Justin Miller wrote: So is there an "RTP_proxy" parameter that can be set for the gateway, or in my case, the user extension? How would I define the separate IP for the RTP stream. -----Original Message----- From: Mathieu Rene [mailto:mrene_lists at avgs.ca] Sent: Wednesday, March 4, 2009 11:47 AM To:freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. Math On 4-Mar-09, at 1:45 PM, Justin Miller wrote: I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060. Thank you Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/f49be49a/attachment-0002.html From dule.maillist at gmail.com Wed Mar 4 15:58:19 2009 From: dule.maillist at gmail.com (Dan Le) Date: Wed, 4 Mar 2009 18:58:19 -0500 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> Message-ID: <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com> What does do exactly? When do you need it? I'll wikify the response. Thanks, Dan On Tue, Feb 10, 2009 at 8:48 PM, Brian West wrote: > I highly recommend you wipe the box/install and install from Scratch > using SVN trunk > > > /b > - Show quoted text - > > On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote: > > > I'm not able to find any documentation on this setting. I think it may > > be newer than my version of FreeSwitch (1.0). What does it do? > > > > Thanks, > > - Jesse > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/880a76ac/attachment-0002.html From mrene_lists at avgs.ca Wed Mar 4 16:00:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 4 Mar 2009 19:00:33 -0500 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com> Message-ID: <226909BF-C3DC-4172-803A-34329C31CC63@avgs.ca> it auto restarts the profile when a network address change is detected. On 4-Mar-09, at 6:58 PM, Dan Le wrote: > What does do exactly? > When do you need it? > > I'll wikify the response. > > Thanks, > Dan > > On Tue, Feb 10, 2009 at 8:48 PM, Brian West > wrote: > I highly recommend you wipe the box/install and install from Scratch > using SVN trunk > > > /b > - Show quoted text - > > On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote: > > > I'm not able to find any documentation on this setting. I think it > may > > be newer than my version of FreeSwitch (1.0). What does it do? > > > > Thanks, > > - Jesse > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/8454d3ba/attachment-0002.html From dule.maillist at gmail.com Wed Mar 4 18:54:50 2009 From: dule.maillist at gmail.com (Dan Le) Date: Wed, 4 Mar 2009 21:54:50 -0500 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <226909BF-C3DC-4172-803A-34329C31CC63@avgs.ca> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com> <226909BF-C3DC-4172-803A-34329C31CC63@avgs.ca> Message-ID: <914fc92a0903041854k4f5a95e2j3692ca09c0582309@mail.gmail.com> Thanks, wiki'd as: http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address Dan On Wed, Mar 4, 2009 at 7:00 PM, Mathieu Rene wrote: > it auto restarts the profile when a network address change is detected.- > Show quoted text - > > On 4-Mar-09, at 6:58 PM, Dan Le wrote: > > What does do exactly? When do > you need it? > I'll wikify the response. > > Thanks, > Dan > > On Tue, Feb 10, 2009 at 8:48 PM, Brian West wrote: > >> I highly recommend you wipe the box/install and install from Scratch >> using SVN trunk >> >> >> /b >> - Show quoted text - >> >> On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote: >> >> > I'm not able to find any documentation on this setting. I think it may >> > be newer than my version of FreeSwitch (1.0). What does it do? >> > >> > Thanks, >> > - Jesse >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/1a0e1975/attachment-0002.html From gerry at pstn2.net Wed Mar 4 19:31:24 2009 From: gerry at pstn2.net (Gerry Hull) Date: Wed, 4 Mar 2009 22:31:24 -0500 Subject: [Freeswitch-users] mod_unistim? Message-ID: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> I hear rumors that someone is porting chan_unistim to mod_unistim for FreeSwitch?? I hope so -- I use this on my Asterisk box and would love to use it with FS. There are TONS of i2004 phones on the surplus market these days... I've been buying NOS i2004s, virgin, for less than $10US... The full duplex speakerphones in these phones are as or better than a Polycom. I'll be happy to test this module -- I'm just not a C/++ guy. Gerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090304/972e1db3/attachment-0002.html From mike at jerris.com Wed Mar 4 19:55:59 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 4 Mar 2009 22:55:59 -0500 Subject: [Freeswitch-users] mod_unistim? In-Reply-To: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> References: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> Message-ID: Due to licensing reasons, you can not "port" a gpl piece of code to FreeSWITCH due to restrictions imposed by the gpl so it is not possible to do this unless all copy-write holders approve a license change. Mike On Mar 4, 2009, at 10:31 PM, Gerry Hull wrote: > I hear rumors that someone is porting chan_unistim to mod_unistim > for FreeSwitch?? I hope so -- I use this on my Asterisk box and > would love to use it with FS. There are TONS of i2004 phones on > the surplus market these days... I've been buying NOS i2004s, > virgin, for less than $10US... The full duplex speakerphones in > these phones are as or better than a Polycom. I'll be happy to > test this module -- I'm just not a C/++ guy. > > Gerry > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Mar 4 20:00:40 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Mar 2009 22:00:40 -0600 Subject: [Freeswitch-users] mod_unistim? In-Reply-To: References: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> Message-ID: Actually in this case you can we were giving FULL rights to do what we wanted with the code from the original author. ;) I still have the emails about it.. and someone asked me about this a few weeks ago. /b On Mar 4, 2009, at 9:55 PM, Michael Jerris wrote: > Due to licensing reasons, you can not "port" a gpl piece of code to > FreeSWITCH due to restrictions imposed by the gpl so it is not > possible to do this unless all copy-write holders approve a license > change. > > Mike From kawarod at laposte.net Thu Mar 5 04:12:51 2009 From: kawarod at laposte.net (rod) Date: Thu, 05 Mar 2009 16:12:51 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID Message-ID: <49AFC1C3.9030603@laposte.net> Dear list, I'd like to rewrite the number in the Remote Party ID header and only in this header. ex: I'd like to prefix the caller ID with a prefix code (000 in this example) in the RPID header : From: Anonymous;tag=1208367 Remote-Party-ID: ;privacy=full;screen=yes;party=calling should become: From: Anonymous;tag=1208367 Remote-Party-ID: ;privacy=full;screen=yes;party=calling But the From field should remain unchanged. And how to strip this prefix: From: Anonymous;tag=1208367 Remote-Party-ID: ;privacy=full;screen=yes;party=calling should become: From: Anonymous;tag=1208367 Remote-Party-ID: ;privacy=full;screen=yes;party=calling regards. From brian at freeswitch.org Thu Mar 5 04:23:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 06:23:02 -0600 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49AFC1C3.9030603@laposte.net> References: <49AFC1C3.9030603@laposte.net> Message-ID: <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> Well this depends on how you're placing the call.. if its a standard bridge you can on the A-Leg set "effective_caller_id_number=000$ {caller_id_number}" before you call bridge. Is the from already in the correct format? /b On Mar 5, 2009, at 6:12 AM, rod wrote: > Dear list, > > I'd like to rewrite the number in the Remote Party ID header and > only in > this header. > > ex: I'd like to prefix the caller ID with a prefix code (000 in this > example) in the RPID header : > > From: Anonymous;tag=1208367 > Remote-Party-ID: > 123456 > @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling > > should become: > From: Anonymous;tag=1208367 > Remote-Party-ID: > 000123456 > @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling > > But the From field should remain unchanged. > > And how to strip this prefix: > From: Anonymous;tag=1208367 > Remote-Party-ID: > 000123456 > @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling > > should become: > From: Anonymous;tag=1208367 > Remote-Party-ID: > 123456 > @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling > > > regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/bd880b43/attachment-0002.html From kawarod at laposte.net Thu Mar 5 04:46:51 2009 From: kawarod at laposte.net (rod) Date: Thu, 05 Mar 2009 16:46:51 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> Message-ID: <49AFC9BB.9090106@laposte.net> Hi Brian, if I use the function effective_caller_id_number with my INVITE, I get this: From: "Anonymous" ;tag=17geyFjX5p0gS. this is not exactly what I'm looking for :p rod Brian West wrote: > Well this depends on how you're placing the call.. if its a standard > bridge you can on the A-Leg set > "effective_caller_id_number=000${caller_id_number}" before you call > bridge. > > Is the from already in the correct format? > > /b > > On Mar 5, 2009, at 6:12 AM, rod wrote: > >> Dear list, >> >> I'd like to rewrite the number in the Remote Party ID header and only in >> this header. >> >> ex: I'd like to prefix the caller ID with a prefix code (000 in this >> example) in the RPID header : >> >> From: Anonymous;tag=1208367 >> Remote-Party-ID: >> ;privacy=full;screen=yes;party=calling >> >> should become: >> From: Anonymous;tag=1208367 >> Remote-Party-ID: >> ;privacy=full;screen=yes;party=calling >> >> But the From field should remain unchanged. >> >> And how to strip this prefix: >> From: Anonymous;tag=1208367 >> Remote-Party-ID: >> ;privacy=full;screen=yes;party=calling >> >> should become: >> From: Anonymous;tag=1208367 >> Remote-Party-ID: >> ;privacy=full;screen=yes;party=calling >> >> >> regards. > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kawarod at laposte.net Thu Mar 5 05:00:50 2009 From: kawarod at laposte.net (rod) Date: Thu, 05 Mar 2009 17:00:50 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49AFC9BB.9090106@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> Message-ID: <49AFCD02.2000603@laposte.net> the A leg invite looks like this: From: "Anonymous" it has been rewritten like this: From: "Anonymous" rod rod wrote: > Hi Brian, > > if I use the function effective_caller_id_number with my INVITE, I get this: > > From: "Anonymous" ;tag=17geyFjX5p0gS. > > this is not exactly what I'm looking for :p > > rod > > > Brian West wrote: > >> Well this depends on how you're placing the call.. if its a standard >> bridge you can on the A-Leg set >> "effective_caller_id_number=000${caller_id_number}" before you call >> bridge. >> >> Is the from already in the correct format? >> >> /b >> >> On Mar 5, 2009, at 6:12 AM, rod wrote: >> >> >>> Dear list, >>> >>> I'd like to rewrite the number in the Remote Party ID header and only in >>> this header. >>> >>> ex: I'd like to prefix the caller ID with a prefix code (000 in this >>> example) in the RPID header : >>> >>> From: Anonymous;tag=1208367 >>> Remote-Party-ID: >>> ;privacy=full;screen=yes;party=calling >>> >>> should become: >>> From: Anonymous;tag=1208367 >>> Remote-Party-ID: >>> ;privacy=full;screen=yes;party=calling >>> >>> But the From field should remain unchanged. >>> >>> And how to strip this prefix: >>> From: Anonymous;tag=1208367 >>> Remote-Party-ID: >>> ;privacy=full;screen=yes;party=calling >>> >>> should become: >>> From: Anonymous;tag=1208367 >>> Remote-Party-ID: >>> ;privacy=full;screen=yes;party=calling >>> >>> >>> regards. >>> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From gibbedhead at gmail.com Thu Mar 5 02:14:57 2009 From: gibbedhead at gmail.com (J Mann/Harry) Date: Thu, 5 Mar 2009 05:14:57 -0500 Subject: [Freeswitch-users] Please end the torment Message-ID: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> No, I've yet to contribute anything, I barely have my system doing what I want. But I REALLY love Freeswitch and I want to see it BURY Asterisk. (Windows server user here) I've been struggling with the XML configs, trying to figure out what does what and where! That's fine, I'm used to it. What I'm NOT used to is the total lack of a forum-based community to join and participate in! Where can users SHARE their configs, help each other, learn from each others mistakes? No DEV forum? I'm speechless. Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" at best! I want stickies, a forum for noobs, converts, a dev forum... So on... "WELCOME To FreeSWITCH!" Am I asking too much here? A FORUM? I can't see how you can spark interest when we're so sorely lacking the most basic and widely used community environments on the net! HELP SOMEBODY? Install SMF ASAP! http://www.simplemachines.org BTW: I hate people who voice problems without offering viable solutions in the process... Disgusting! So if someone can offer up a simple hosting account, Control Panel 10, Windows Linux whatever... I'll be more than happy to have SMF setup and receiving user registrations within 24 hours! I've done it dozens of times before. I will then gladly turn over the keys to the kingdom to the "powers that be" and take a backseat, being simply a happy user from that point on! Please folks, please. I'm dying over here and I'm sure I'm not alone! I'm searching Google and finding nothing!! FORUMS! Harry (email me here) switchserver at gmail.com (my FS email) From mrene_lists at avgs.ca Thu Mar 5 06:04:15 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Mar 2009 09:04:15 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> Message-ID: The wiki explains all that and allows all that, whats wrong with IRC? Come and ask questions if you don't understand, you'll get your answers quicker than complaining about the lack of forum. Math On 5-Mar-09, at 5:14 AM, J Mann/Harry wrote: > No, I've yet to contribute anything, I barely have my system doing > what I want. But I REALLY love Freeswitch and I want to see it BURY > Asterisk. (Windows server user here) > > I've been struggling with the XML configs, trying to figure out what > does what and where! That's fine, I'm used to it. What I'm NOT used to > is the total lack of a forum-based community to join and participate > in! Where can users SHARE their configs, help each other, learn from > each others mistakes? No DEV forum? I'm speechless. > > Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" > at best! I want stickies, a forum for noobs, converts, a dev forum... > So on... > > "WELCOME To FreeSWITCH!" > > Am I asking too much here? A FORUM? > > I can't see how you can spark interest when we're so sorely lacking > the most basic and widely used community environments on the net! > > HELP SOMEBODY? Install SMF ASAP! > http://www.simplemachines.org > > BTW: I hate people who voice problems without offering viable > solutions in the process... Disgusting! So if someone can offer up a > simple hosting account, Control Panel 10, Windows Linux whatever... > I'll be more than happy to have SMF setup and receiving user > registrations within 24 hours! I've done it dozens of times before. I > will then gladly turn over the keys to the kingdom to the "powers that > be" and take a backseat, being simply a happy user from that point on! > > Please folks, please. I'm dying over here and I'm sure I'm not alone! > I'm searching Google and finding nothing!! FORUMS! > > Harry (email me here) > switchserver at gmail.com (my FS email) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at suspicious.org Thu Mar 5 06:11:19 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 05 Mar 2009 08:11:19 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: Message-ID: Not to mention there is a full archive of the mailing list on nabble http://www.nabble.com/Freeswitch-users-f32209.html > From: Mathieu Rene > Reply-To: > Date: Thu, 5 Mar 2009 09:04:15 -0500 > To: > Subject: Re: [Freeswitch-users] Please end the torment > > The wiki explains all that and allows all that, whats wrong with IRC? > Come and ask questions if you don't understand, you'll get your > answers quicker than complaining about the lack of forum. > > Math > > On 5-Mar-09, at 5:14 AM, J Mann/Harry wrote: > >> No, I've yet to contribute anything, I barely have my system doing >> what I want. But I REALLY love Freeswitch and I want to see it BURY >> Asterisk. (Windows server user here) >> >> I've been struggling with the XML configs, trying to figure out what >> does what and where! That's fine, I'm used to it. What I'm NOT used to >> is the total lack of a forum-based community to join and participate >> in! Where can users SHARE their configs, help each other, learn from >> each others mistakes? No DEV forum? I'm speechless. >> >> Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" >> at best! I want stickies, a forum for noobs, converts, a dev forum... >> So on... >> >> "WELCOME To FreeSWITCH!" >> >> Am I asking too much here? A FORUM? >> >> I can't see how you can spark interest when we're so sorely lacking >> the most basic and widely used community environments on the net! >> >> HELP SOMEBODY? Install SMF ASAP! >> http://www.simplemachines.org >> >> BTW: I hate people who voice problems without offering viable >> solutions in the process... Disgusting! So if someone can offer up a >> simple hosting account, Control Panel 10, Windows Linux whatever... >> I'll be more than happy to have SMF setup and receiving user >> registrations within 24 hours! I've done it dozens of times before. I >> will then gladly turn over the keys to the kingdom to the "powers that >> be" and take a backseat, being simply a happy user from that point on! >> >> Please folks, please. I'm dying over here and I'm sure I'm not alone! >> I'm searching Google and finding nothing!! FORUMS! >> >> Harry (email me here) >> switchserver at gmail.com (my FS email) >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Mar 5 06:22:32 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Mar 2009 06:22:32 -0800 (PST) Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> Message-ID: <1236262952915-2429661.post@n2.nabble.com> Much more than an archive, nabble makes a full embeddable forum that is linked to the mailing list. We will be embedding this in our webpage soon for the best of both worlds, a forum and a mailing list without the additional overhead of having to monitor 2 things. Mike Ken Rice-3 wrote: > > Not to mention there is a full archive of the mailing list on nabble > > http://www.nabble.com/Freeswitch-users-f32209.html > > -- View this message in context: http://n2.nabble.com/Please-end-the-torment-tp2429589p2429661.html Sent from the freeswitch-users mailing list archive at Nabble.com. From damin at nacs.net Thu Mar 5 06:46:44 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 5 Mar 2009 09:46:44 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: Message-ID: <123301c99da1$33ee98e0$9bcbcaa0$@net> I hate forums. Forums suck. They are a pain in the ass to search and find things. I prefer the mailing list and the Wiki. Can we please keep it that way? It is really easy to find stuff via Nabble and the Wiki. From gmaruzz at celliax.org Thu Mar 5 07:11:25 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 5 Mar 2009 16:11:25 +0100 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <1236262952915-2429661.post@n2.nabble.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <1236262952915-2429661.post@n2.nabble.com> Message-ID: <7b197bef0903050711v5a8d0eefo31c9e476fe6167ae@mail.gmail.com> On Thu, Mar 5, 2009 at 3:22 PM, Michael Jerris wrote: > > Much more than an archive, nabble makes a full embeddable forum that is > linked to the mailing list. ?We will be embedding this in our webpage soon > for the best of both worlds, a forum and a mailing list without the > additional overhead of having to monitor 2 things. agree! From helmut.kuper at ewetel.de Thu Mar 5 07:31:34 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 05 Mar 2009 16:31:34 +0100 Subject: [Freeswitch-users] Patch for mod_event_socket to fire real CUSTOM events via sendevent command Message-ID: <49AFF056.70706@ewetel.de> Hello, I enhanced mod_event_socket's sendevent command to fire real CUSTOM events with correct subclass and trailing custom headers, so that subscribed nodes can receive those events. Is there a way to get the patch into trunk? regards Helmut From csorlie at teldio.com Thu Mar 5 07:41:28 2009 From: csorlie at teldio.com (Cameron Sorlie) Date: Thu, 5 Mar 2009 10:41:28 -0500 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD Message-ID: In a sense, you might say I did futz with mod_vmd ... to create mod_vad. There appeared to be just no (easy) way to modify the internal VAD code in the FreeSWITCH core (see switch_rtp.c) to identify the origins of voice activity. And rather than build into mod_vmd, which is a special purpose tool, a separate module for VAD seemed like a reasonable idea. In short, the mod_vad which I've written up independently monitors the read and the write legs of the session it is run on, and tags each VAD_TALK and VAD_NOTALK event it fires with a user-supplied identification marker (a short string) for the leg which the event relates to. At the moment, the VAD algorithm is dead simple, and is much like the one in the core. I will be happy to submit this module, in a little while, after I've had a chance to make it perhaps a bit more useable outside of our own application context. Cam On Mon, Mar 2, 2009 at 5:43 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > > On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale > wrote: > > i think that's what mod_vmd does > > > I think that's right. It just does the opposite - instead of looking > for differing power levels it looks for the same power level. In other > words it tries to detect distinctly non-human sound. I'll bet you > could futz with that code and have it fire off events when it detects > what it believes is human speech. > > -MC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/f240a1b3/attachment-0002.html From mrene_lists at avgs.ca Thu Mar 5 07:41:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Mar 2009 10:41:33 -0500 Subject: [Freeswitch-users] Patch for mod_event_socket to fire real CUSTOM events via sendevent command In-Reply-To: <49AFF056.70706@ewetel.de> References: <49AFF056.70706@ewetel.de> Message-ID: <8BC75FF7-C82D-49E4-BFEA-E40887372A57@avgs.ca> Sure, open a JIRA as a improvement, and prefix your bug name with [patch] http://jira.freeswitch.org/ Math On 5-Mar-09, at 10:31 AM, Helmut Kuper wrote: > Hello, > > I enhanced mod_event_socket's sendevent command to fire real CUSTOM > events with correct subclass and trailing custom headers, so that > subscribed nodes can receive those events. Is there a way to get the > patch into trunk? > > regards > Helmut > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Mar 5 07:51:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 09:51:15 -0600 Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD In-Reply-To: References: Message-ID: <9AE4AAC4-34A3-4758-B64E-38B0056C5F30@freeswitch.org> Kewl you can contribute via http://jira.freeswitch.org ;) /b On Mar 5, 2009, at 9:41 AM, Cameron Sorlie wrote: > In a sense, you might say I did futz with mod_vmd ... to create > mod_vad. There appeared to be just no (easy) way to modify the > internal VAD code in the FreeSWITCH core (see switch_rtp.c) to > identify the origins of voice activity. And rather than build into > mod_vmd, which is a special purpose tool, a separate module for VAD > seemed like a reasonable idea. > > In short, the mod_vad which I've written up independently monitors > the read and the write legs of the session it is run on, and tags > each VAD_TALK and VAD_NOTALK event it fires with a user-supplied > identification marker (a short string) for the leg which the event > relates to. At the moment, the VAD algorithm is dead simple, and is > much like the one in the core. I will be happy to submit this > module, in a little while, after I've had a chance to make it > perhaps a bit more useable outside of our own application context. > > Cam From helmut.kuper at ewetel.de Thu Mar 5 07:52:27 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 05 Mar 2009 16:52:27 +0100 Subject: [Freeswitch-users] Problems with record_stereo In-Reply-To: References: <49A7F393.6080406@ewetel.de> <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com> <49ACE742.5090809@ewetel.de> <49AD4BAA.8090208@ewetel.de> <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org> <49AE7404.8000905@ewetel.de> <49AE7D43.9070409@ewetel.de> <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org> <49AEA042.3090908@ewetel.de> Message-ID: <49AFF53B.5000802@ewetel.de> Hello Brian, I checked the files without earphones and with Audacity. You are right Brian, everything is fine. Sorry for the inconvenience. Thx for your patience. regards helmut From BenHoltsclaw at averyschools.net Thu Mar 5 09:03:50 2009 From: BenHoltsclaw at averyschools.net (Ben Holtsclaw) Date: Thu, 05 Mar 2009 12:03:50 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> Message-ID: <49AFBFA6.45B7.0079.0@averyschools.net> I agree with Harry. I do not like the mailing list. Those that do like the mailing list always advocate Nabble. For those that advocate that solution, do you even realize that you can't post on Nabble unless you are subscribed to the mailing list? I am also not a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client at the same time I uninstalled my NNTP reader. Most of the time, I actually find it difficult to obtain support in the #freeswitch channel. Once you ask the question, if somebody doesn't happen to be there that knows the answer, then you're screwed. How many times have I asked a question only to wait 30 seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the question again? I have found the conversation in #openzap to be much more focused. Thank goodness I'm using that module! In that channel, I never see conversations about cd burners, somebody's girlfriend in South America, or off color jokes about someone's sexual proclivity. And because I know I'll get flamed for saying that, just look at this: [23:10] <{tasker}> me, too, but i'm a different animal [23:10] <{tasker}> in NY and in Miami i went nutz [23:10] lol [23:10] * jefferai is now known as lollerai [23:10] yeah i love her [23:10] <{tasker}> latinas everywhere [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left #freeswitch [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed out) ) [23:11] here its blond blue eyed girls [23:11] * lollerai is now known as lolferai [23:11] brazilians... hopefully she's hot. i've seen some pretty dodgy looking chicks from there [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, "yeaaaaaaaaah, it's just for a few days" [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch [23:11] <{tasker}> blonde / blue eyes are overrated [23:11] <{tasker}> give me a latina any day [23:11] best thing around here though If I'm going into #freeswitch at 11pm at night, it's probably because I really need some help with some problem I've run into after hours. Can you imagine me injecting a question about a SIP profile into that conversation?? ALL that aside... I'm willing to use a carrier pigeon if that's the way the three primary developers wish to communicate. They have been instrumental in getting my project where it is today. You know the saying... beggars can't be choosers. Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x2301 >>> On 3/5/2009 at 5:14 AM, "J Mann/Harry" wrote: No, I've yet to contribute anything, I barely have my system doing what I want. But I REALLY love Freeswitch and I want to see it BURY Asterisk. (Windows server user here) I've been struggling with the XML configs, trying to figure out what does what and where! That's fine, I'm used to it. What I'm NOT used to is the total lack of a forum-based community to join and participate in! Where can users SHARE their configs, help each other, learn from each others mistakes? No DEV forum? I'm speechless. Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" at best! I want stickies, a forum for noobs, converts, a dev forum... So on... "WELCOME To FreeSWITCH!" Am I asking too much here? A FORUM? I can't see how you can spark interest when we're so sorely lacking the most basic and widely used community environments on the net! HELP SOMEBODY? Install SMF ASAP! http://www.simplemachines.org BTW: I hate people who voice problems without offering viable solutions in the process... Disgusting! So if someone can offer up a simple hosting account, Control Panel 10, Windows Linux whatever... I'll be more than happy to have SMF setup and receiving user registrations within 24 hours! I've done it dozens of times before. I will then gladly turn over the keys to the kingdom to the "powers that be" and take a backseat, being simply a happy user from that point on! Please folks, please. I'm dying over here and I'm sure I'm not alone! I'm searching Google and finding nothing!! FORUMS! Harry (email me here) switchserver at gmail.com (my FS email) _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/783d1083/attachment-0002.html From brian at freeswitch.org Thu Mar 5 09:16:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 11:16:08 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <49AFBFA6.45B7.0079.0@averyschools.net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: I have been trying to push all the social talk into #freeswitch-social to keep #freeswitch on topic.. sometimes after hours in the US it gets a bit off topic. I'm usually alive in the channel till around 11PM+ CST most days. I take questions and answer questions at all hours if I'm awake... I too am guilty of going off topic. /b On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: > I agree with Harry. I do not like the mailing list. Those that do > like the mailing list always advocate Nabble. For those that > advocate that solution, do you even realize that you can't post on > Nabble unless you are subscribed to the mailing list? I am also not > a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in > years! I think I uninstalled my IRC client at the same time I > uninstalled my NNTP reader. Most of the time, I actually find it > difficult to obtain support in the #freeswitch channel. Once you ask > the question, if somebody doesn't happen to be there that knows the > answer, then you're screwed. How many times have I asked a question > only to wait 30 seconds and then see, "anthm has joined > #freeswitch." Crap...do I ask the question again? I have found the > conversation in #openzap to be much more focused. Thank goodness I'm > using that module! In that channel, I never see conversations about > cd burners, somebody's girlfriend in South America, or off color > jokes about someone's sexual proclivity. And because I know I'll get > flamed for saying that, just look at this: > > [23:10] <{tasker}> me, too, but i'm a different animal > [23:10] <{tasker}> in NY and in Miami i went nutz > [23:10] lol > [23:10] * jefferai is now known as lollerai > [23:10] yeah i love her > [23:10] <{tasker}> latinas everywhere > [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left > #freeswitch > [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection > timed out)) > [23:11] here its blond blue eyed girls > [23:11] * lollerai is now known as lolferai > [23:11] brazilians... hopefully she's hot. i've seen some > pretty dodgy looking chicks from there > [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, > "yeaaaaaaaaah, it's just for a few days" > [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though > > If I'm going into #freeswitch at 11pm at night, it's probably > because I really need some help with some problem I've run into > after hours. Can you imagine me injecting a question about a SIP > profile into that conversation?? > > ALL that aside... I'm willing to use a carrier pigeon if that's the > way the three primary developers wish to communicate. They have been > instrumental in getting my project where it is today. You know the > saying... beggars can't be choosers. > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/da576a03/attachment-0002.html From mrene_lists at avgs.ca Thu Mar 5 09:19:31 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Mar 2009 12:19:31 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: The bot even takes people's question and keeps them in list so when someone that knows shows up he can answer it, thats the ~take-a-number option... On 5-Mar-09, at 12:16 PM, Brian West wrote: > I have been trying to push all the social talk into #freeswitch- > social to keep #freeswitch on topic.. sometimes after hours in the > US it gets a bit off topic. I'm usually alive in the channel till > around 11PM+ CST most days. I take questions and answer questions > at all hours if I'm awake... I too am guilty of going off topic. > > /b > > > On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: > >> I agree with Harry. I do not like the mailing list. Those that do >> like the mailing list always advocate Nabble. For those that >> advocate that solution, do you even realize that you can't post on >> Nabble unless you are subscribed to the mailing list? I am also not >> a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in >> years! I think I uninstalled my IRC client at the same time I >> uninstalled my NNTP reader. Most of the time, I actually find it >> difficult to obtain support in the #freeswitch channel. Once you >> ask the question, if somebody doesn't happen to be there that knows >> the answer, then you're screwed. How many times have I asked a >> question only to wait 30 seconds and then see, "anthm has joined >> #freeswitch." Crap...do I ask the question again? I have found the >> conversation in #openzap to be much more focused. Thank goodness >> I'm using that module! In that channel, I never see conversations >> about cd burners, somebody's girlfriend in South America, or off >> color jokes about someone's sexual proclivity. And because I know >> I'll get flamed for saying that, just look at this: >> >> [23:10] <{tasker}> me, too, but i'm a different animal >> [23:10] <{tasker}> in NY and in Miami i went nutz >> [23:10] lol >> [23:10] * jefferai is now known as lollerai >> [23:10] yeah i love her >> [23:10] <{tasker}> latinas everywhere >> [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has >> left #freeswitch >> [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection >> timed out)) >> [23:11] here its blond blue eyed girls >> [23:11] * lollerai is now known as lolferai >> [23:11] brazilians... hopefully she's hot. i've seen some >> pretty dodgy looking chicks from there >> [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, >> "yeaaaaaaaaah, it's just for a few days" >> [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch >> [23:11] <{tasker}> blonde / blue eyes are overrated >> [23:11] <{tasker}> give me a latina any day >> [23:11] best thing around here though >> >> If I'm going into #freeswitch at 11pm at night, it's probably >> because I really need some help with some problem I've run into >> after hours. Can you imagine me injecting a question about a SIP >> profile into that conversation?? >> >> ALL that aside... I'm willing to use a carrier pigeon if that's the >> way the three primary developers wish to communicate. They have >> been instrumental in getting my project where it is today. You know >> the saying... beggars can't be choosers. >> >> >> Ben Holtsclaw >> Network Engineer >> Avery County Schools >> Ph: 828.733.3567 x2301 >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/de361bcb/attachment-0002.html From egghunt at gmail.com Thu Mar 5 09:27:59 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Thu, 5 Mar 2009 14:27:59 -0300 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: Besides #freeswitch-social and #openzap, there are other channels related to the project. List: http://wiki.freeswitch.org/wiki/Main_Page#Community_and_Support On Thu, Mar 5, 2009 at 2:19 PM, Mathieu Rene wrote: > The bot even takes people's question and keeps them in list so when someone > that knows shows up he can answer it, thats the ~take-a-number option... > On 5-Mar-09, at 12:16 PM, Brian West wrote: > > I have been trying to push all the social talk into #freeswitch-social to > keep #freeswitch on topic.. sometimes after hours in the US it gets a bit > off topic. I'm usually alive in the channel till around 11PM+ CST most > days. I take questions and answer questions at all hours if I'm awake... I > too am guilty of going off topic. > /b > > > On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: > > I agree with Harry. I do not like the mailing list. Those that do like the > mailing list always advocate Nabble. For those that advocate that solution, > do you even realize that you can't post on Nabble unless you *are* subscribed > to the mailing list? I am also not a fan of IRC. Before I came upon > FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client > at the same time I uninstalled my NNTP reader. Most of the time, I actually > find it difficult to obtain support in the #freeswitch channel. Once you ask > the question, if somebody doesn't happen to be there that knows the answer, > then you're screwed. How many times have I asked a question only to wait 30 > seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the > question again? I *have* found the conversation in #openzap to be much > more focused. Thank goodness I'm using that module! In that channel, I never > see conversations about cd burners, somebody's girlfriend in South America, > or off color jokes about someone's sexual proclivity. And because I know > I'll get flamed for saying that, just look at this: > > [23:10] <{tasker}> me, too, but i'm a different animal > [23:10] <{tasker}> in NY and in Miami i went nutz > [23:10] lol > [23:10] * jefferai is now known as lollerai > [23:10] yeah i love her > [23:10] <{tasker}> latinas everywhere > [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left > #freeswitch > [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed > out) ) > [23:11] here its blond blue eyed girls > [23:11] * lollerai is now known as lolferai > [23:11] brazilians... hopefully she's hot. i've seen some pretty > dodgy looking chicks from there > [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, > "yeaaaaaaaaah, it's just for a few days" > [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though > > If I'm going into #freeswitch at 11pm at night, it's probably because I > really need some help with some problem I've run into after hours. Can you > imagine me injecting a question about a SIP profile into that conversation?? > > ALL that aside... I'm willing to use a carrier pigeon if that's the way the > three primary developers wish to communicate. They have been instrumental in > getting my project where it is today. You know the saying... beggars can't > be choosers. > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Arnaldo M Pereira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/9f412589/attachment-0002.html From BenHoltsclaw at averyschools.net Thu Mar 5 09:27:35 2009 From: BenHoltsclaw at averyschools.net (Ben Holtsclaw) Date: Thu, 05 Mar 2009 12:27:35 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: <49AFC537.45B7.0079.0@averyschools.net> The problem with take-a-number is what if I'm not there when someone can answer it? >>> On 3/5/2009 at 12:19 PM, Mathieu Rene wrote: The bot even takes people's question and keeps them in list so when someone that knows shows up he can answer it, thats the ~take-a-number option... On 5-Mar-09, at 12:16 PM, Brian West wrote: I have been trying to push all the social talk into #freeswitch-social to keep #freeswitch on topic.. sometimes after hours in the US it gets a bit off topic. I'm usually alive in the channel till around 11PM+ CST most days. I take questions and answer questions at all hours if I'm awake... I too am guilty of going off topic. /b On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: I agree with Harry. I do not like the mailing list. Those that do like the mailing list always advocate Nabble. For those that advocate that solution, do you even realize that you can't post on Nabble unless you are subscribed to the mailing list? I am also not a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client at the same time I uninstalled my NNTP reader. Most of the time, I actually find it difficult to obtain support in the #freeswitch channel. Once you ask the question, if somebody doesn't happen to be there that knows the answer, then you're screwed. How many times have I asked a question only to wait 30 seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the question again? I have found the conversation in #openzap to be much more focused. Thank goodness I'm using that module! In that channel, I never see conversations about cd burners, somebody's girlfriend in South America, or off color jokes about someone's sexual proclivity. And because I know I'll get flamed for saying that, just look at this: [23:10] <{tasker}> me, too, but i'm a different animal [23:10] <{tasker}> in NY and in Miami i went nutz [23:10] lol [23:10] * jefferai is now known as lollerai [23:10] yeah i love her [23:10] <{tasker}> latinas everywhere [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left #freeswitch [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed out)) [23:11] here its blond blue eyed girls [23:11] * lollerai is now known as lolferai [23:11] brazilians... hopefully she's hot. i've seen some pretty dodgy looking chicks from there [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, "yeaaaaaaaaah, it's just for a few days" [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch [23:11] <{tasker}> blonde / blue eyes are overrated [23:11] <{tasker}> give me a latina any day [23:11] best thing around here though If I'm going into #freeswitch at 11pm at night, it's probably because I really need some help with some problem I've run into after hours. Can you imagine me injecting a question about a SIP profile into that conversation?? ALL that aside... I'm willing to use a carrier pigeon if that's the way the three primary developers wish to communicate. They have been instrumental in getting my project where it is today. You know the saying... beggars can't be choosers. Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x2301 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/daae766e/attachment-0002.html From gerry at pstn2.net Thu Mar 5 09:30:21 2009 From: gerry at pstn2.net (Gerry Hull) Date: Thu, 5 Mar 2009 12:30:21 -0500 Subject: [Freeswitch-users] mod_unistim? In-Reply-To: References: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com> Message-ID: <98a86adf0903050930t7b0df79fm5524a3ec0aaa3b4f@mail.gmail.com> Yeah, I was looking at the Developers IRC log... I see that milkj has given you an MPL license... Has anyone started development on the port? On Wed, Mar 4, 2009 at 11:00 PM, Brian West wrote: > Actually in this case you can we were giving FULL rights to do what we > wanted with the code from the original author. ;) I still have the > emails about it.. and someone asked me about this a few weeks ago. > > /b > > On Mar 4, 2009, at 9:55 PM, Michael Jerris wrote: > > > Due to licensing reasons, you can not "port" a gpl piece of code to > > FreeSWITCH due to restrictions imposed by the gpl so it is not > > possible to do this unless all copy-write holders approve a license > > change. > > > > Mike > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/68c7cea4/attachment-0002.html From mike at jerris.com Thu Mar 5 09:31:14 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Mar 2009 12:31:14 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <49AFBFA6.45B7.0079.0@averyschools.net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: <6F7B8F0B-0975-452D-A6C1-AE484131E2AB@jerris.com> On Mar 5, 2009, at 12:03 PM, Ben Holtsclaw wrote: > I agree with Harry. I do not like the mailing list. Those that do > like the mailing list always advocate Nabble. For those that > advocate that solution, do you even realize that you can't post on > Nabble unless you are subscribed to the mailing list? Fair points, the plan to address this is that we are moving to a new back-end to our hosting infrastructure that has unified logins. Once this is in place we can make it so you are a "member" of the mailing list as soon as you have an id, although set to not receive mail, then we'll have a subscriptions options page where you can just check the lists you want sent to you via email, and the nabble forums can be embedded in our page directly. It is not perfect but much closer to something usable. I understand that some are more comfortable than others with different modes of communication. We are trying to find a balance between being able to serve the community with the best way to communicate for them, and us being able to actually monitor and maintain it. A forum is pretty useless if no one responds or monitors it and already we are flooded by the amount on the mailing list, having to answer the same question the same day in both because someone prefers one to the other and having to spend time checking both is not a good use of time. Do you think that if we better integrate the nabble forums into our site and list subscriptions that it would be a usable system? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/5b2f6de1/attachment-0002.html From anthony.minessale at gmail.com Thu Mar 5 09:33:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Mar 2009 11:33:03 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: <191c3a030903050933p6df4a725o9be751a390f9c916@mail.gmail.com> Ok, SO? We should make a forum? See this one? http://www.voip-info.org/boards/index.php?b=6 Yah, that one has been around for many months and nobody maintains it. We do have a link to it on our homepage but nobody reads it so, there is a dilemma here right? IRC is for sure more useful of a resource between 9am and 6pm GMT-6 in the timezone where me and bkw live. Any other times it would be up to other community members like the few who have been stepping up in a major way in the last few weeks, thank you all. I don't know how good of a job we are doing, but we are a small group and we already spend most of every day managing a bug tracker, irc, and this list. If you took a poll 8/10 people hate forums. I don't really know if i like them or not, I have enjoyed finding forums before that discussed something that happened ages ago that relates to a problem I have and being able to skim months into the future reading the end results, but more times than not they go south and contain a bunch of flaming back and forth about midway through. We already need voulenteers to help manage the resources we already have, Website, Mailing list, JIRA. If we introduce more we, run the risk of doing a bad job at maintaining them. Is anyone willing to help? If you can use nabble to post as long as you have an account on the mailing list, can't you just disable mail dilevery and use the nabble exclusively? If not is there another solution? I really need some help in this department to even consider introducing any more community resources. On Thu, Mar 5, 2009 at 11:16 AM, Brian West wrote: > I have been trying to push all the social talk into #freeswitch-social to > keep #freeswitch on topic.. sometimes after hours in the US it gets a bit > off topic. I'm usually alive in the channel till around 11PM+ CST most > days. I take questions and answer questions at all hours if I'm awake... I > too am guilty of going off topic. > /b > > > On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote: > > I agree with Harry. I do not like the mailing list. Those that do like the > mailing list always advocate Nabble. For those that advocate that solution, > do you even realize that you can't post on Nabble unless you *are* subscribed > to the mailing list? I am also not a fan of IRC. Before I came upon > FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client > at the same time I uninstalled my NNTP reader. Most of the time, I actually > find it difficult to obtain support in the #freeswitch channel. Once you ask > the question, if somebody doesn't happen to be there that knows the answer, > then you're screwed. How many times have I asked a question only to wait 30 > seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the > question again? I *have* found the conversation in #openzap to be much > more focused. Thank goodness I'm using that module! In that channel, I never > see conversations about cd burners, somebody's girlfriend in South America, > or off color jokes about someone's sexual proclivity. And because I know > I'll get flamed for saying that, just look at this: > > [23:10] <{tasker}> me, too, but i'm a different animal > [23:10] <{tasker}> in NY and in Miami i went nutz > [23:10] lol > [23:10] * jefferai is now known as lollerai > [23:10] yeah i love her > [23:10] <{tasker}> latinas everywhere > [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left > #freeswitch > [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed > out) ) > [23:11] here its blond blue eyed girls > [23:11] * lollerai is now known as lolferai > [23:11] brazilians... hopefully she's hot. i've seen some pretty > dodgy looking chicks from there > [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, > "yeaaaaaaaaah, it's just for a few days" > [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though > > If I'm going into #freeswitch at 11pm at night, it's probably because I > really need some help with some problem I've run into after hours. Can you > imagine me injecting a question about a SIP profile into that conversation?? > > ALL that aside... I'm willing to use a carrier pigeon if that's the way the > three primary developers wish to communicate. They have been instrumental in > getting my project where it is today. You know the saying... beggars can't > be choosers. > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/2d4752ed/attachment-0002.html From brian at freeswitch.org Thu Mar 5 09:37:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 11:37:36 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <49AFC537.45B7.0079.0@averyschools.net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> Message-ID: <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> Well you can always shoot an email to the mailing list... or wait for someone that can answer it.. which reminds me we really need people to volunteer to help out on the IRC channel so we have skilled people able to answer questions 24/7, I personally cover 12-16 hours a day most of the time. Even weekends! I was even up at 4am today and answered a few. /b On Mar 5, 2009, at 11:27 AM, Ben Holtsclaw wrote: > The problem with take-a-number is what if I'm not there when someone > can answer it? From egghunt at gmail.com Thu Mar 5 09:43:04 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Thu, 5 Mar 2009 14:43:04 -0300 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> Message-ID: On Thu, Mar 5, 2009 at 2:37 PM, Brian West wrote: > Well you can always shoot an email to the mailing list... or wait for > someone that can answer it.. which reminds me we really need people to > volunteer to help out on the IRC channel so we have skilled people > able to answer questions 24/7, I personally cover 12-16 hours a day > most of the time. Even weekends! I was even up at 4am today and > answered a few. I'm usually on the channel starting 9am (GMT -3). Can't answer every question like you guys, but I can surely help. > > > /b > > On Mar 5, 2009, at 11:27 AM, Ben Holtsclaw wrote: > > > The problem with take-a-number is what if I'm not there when someone > > can answer it? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Arnaldo M Pereira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/72df5d9c/attachment-0002.html From damin at nacs.net Thu Mar 5 09:43:43 2009 From: damin at nacs.net (Gregory Boehnlein) Date: Thu, 5 Mar 2009 12:43:43 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> Message-ID: <131e01c99db9$ed440ba0$c7cc22e0$@net> You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :) From Prometheus001 at gmx.net Thu Mar 5 12:20:18 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 05 Mar 2009 21:20:18 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> <49ADD3E4.20408@gmx.net> <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> Message-ID: <49B03402.8050601@gmx.net> Hello Brian, concerning > Well you should use ESL then ;) I simply do not understand what you mean by this. Is it sarcastic? Am I asking stupid questions? After upgrading Freeswitch to the newest trunk, mod_pocketsphinx didn't work anymore. So I asked this mailing list about information about what happened. I understand now that there were some significant changes in mod_pocketsphinx and that also some other files have to be updated. I could not find any documentation about these changes, and asking here on this mailing list was rather disappointing for me. Some bits, yes. Some things don't work/crash, as I have read here. We are not using Freeswitch just as a toy to play around. Sometimes it's simply important to know which impact a certain change may have on our system. And other people will run into the same problem. So any advice was needed about the status and how to make it work. I'll update the wiki with this information (as I usually do), I promise. I honor the great work you do and freeswitch is really great. But asking: >Will there also be major changes in the events I receive through mod_eventsocket? >Will I need updated grammar files for the other models too? and receiving > Well you should use ESL then ;) is frustrating. Best regards Peter Brian West schrieb: > Well you should use ESL then ;) > > /b > > On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote: > > >> Thank you Brian, >> >> I will try this later. >> >> Currently I was happy to get this working on SVN 10003. >> >> As mod_pockesphinx has changed/evolved significantely: Will there also >> be major changes in the events I receive through mod_eventsocket? >> I spend some time on parsing the right data out of the eventsocket >> interface, and I would just have an idea, if I will have to expect >> significant work to do, when I later switch to the current SVN. >> >> Will I need updated grammar files for the other models too? >> >> Best regards >> Peter >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu Mar 5 12:32:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Mar 2009 12:32:29 -0800 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49B03402.8050601@gmx.net> References: <49AC6EE4.9080509@gmx.net> <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com> <49AD0F7B.7000802@gmx.net> <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org> <49ADD3E4.20408@gmx.net> <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org> <49B03402.8050601@gmx.net> Message-ID: <87f2f3b90903051232l22db660blba57c46208f42d63@mail.gmail.com> On Thu, Mar 5, 2009 at 12:20 PM, Peter P GMX wrote: > Hello Brian, > > concerning >> Well you should use ESL then ;) > I simply do not understand what you mean by this. Is it sarcastic? Am I > asking stupid questions? > ESL = Event Socket Library. It is an abstraction layer to make interacting with the FS event socket a little easier. Look in the source directory under libs/esl and you'll see all sorts of stuff. Also check out the new-but-growing ESL wiki page: http://wiki.freeswitch.org/wiki/Esl -MC From kristian.kielhofner at gmail.com Thu Mar 5 12:39:27 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 5 Mar 2009 15:39:27 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <131e01c99db9$ed440ba0$c7cc22e0$@net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> <131e01c99db9$ed440ba0$c7cc22e0$@net> Message-ID: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> A bunch of telephony geeks and a 1900 number - what could go wrong? Anyways, I too don't understand why people prefer forums. I follow dozens of mailling lists and a half a dozen e-mail addresses without ever leaving my mail client. My mail client happens to be gmail, btw: - Much more customization, filtering, etc possible than any "web forum" - "Local" copies of all messages - Search is awesome, ever hear of Google? ;) Web forums are good when you have to serve ads to people to get paid. Other than that they are certainly not the ideal tool for the job. Besides (and don't take this as an insult) - have you ever compared the web forums to the mailing lists for projects that offer both? Say what you want to say about mailing lists and IRC but the reality (usually) is the l33tz all hang out here and web forums (almost always) end up with the same groups of n00bz circling around and around trying to figure out how to accomplish even the most basic of tasks. Obviously that can go both ways but as a rule of thumb the people that are usually in a position to help others typically prefer mailing lists (probably for some of the reasons I cited above). Or maybe they are just old gray hairs too stuck in their ways. I don't know. ;) On Thu, Mar 5, 2009 at 12:43 PM, Gregory Boehnlein wrote: > You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :) > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From mike at jerris.com Thu Mar 5 13:57:30 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Mar 2009 16:57:30 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <49AFC537.45B7.0079.0@averyschools.net> <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org> <131e01c99db9$ed440ba0$c7cc22e0$@net> <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> Message-ID: <5C6F6B6E-EE30-44CD-996D-ED0A2FD0F894@jerris.com> On Mar 5, 2009, at 3:39 PM, Kristian Kielhofner wrote: > A bunch of telephony geeks and a 1900 number - what could go wrong? > > Anyways, I too don't understand why people prefer forums. > > I follow dozens of mailling lists and a half a dozen e-mail addresses > without ever leaving my mail client. My mail client happens to be > gmail, btw: > > - Much more customization, filtering, etc possible than any "web > forum" > - "Local" copies of all messages > - Search is awesome, ever hear of Google? ;) > > Web forums are good when you have to serve ads to people to get > paid. Other than that they are certainly not the ideal tool for the > job. > > Besides (and don't take this as an insult) - have you ever compared > the web forums to the mailing lists for projects that offer both? Say > what you want to say about mailing lists and IRC but the reality > (usually) is the l33tz all hang out here and web forums (almost > always) end up with the same groups of n00bz circling around and > around trying to figure out how to accomplish even the most basic of > tasks. > > Obviously that can go both ways but as a rule of thumb the people > that are usually in a position to help others typically prefer mailing > lists (probably for some of the reasons I cited above). Or maybe they > are just old gray hairs too stuck in their ways. I don't know. ;) I previously thought the same about the insiders always preferring mailing lists but I have a friend who is a core member of the cacti developers and they apparently prefer the forums to the lists, but have both. Food for thought at least. MIke From dynaguy at gmail.com Thu Mar 5 12:19:49 2009 From: dynaguy at gmail.com (Dyna Guy) Date: Thu, 5 Mar 2009 12:19:49 -0800 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot Message-ID: I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh install) and followed the instruction on the wiki installed the new Freeswitch v1.0.3. After the installation I can start FS by issus command "/usr/local/freeswitch/bin/freeswitch". After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and chmod it to 755. Reboot the server and FS is not running. If I try the start up script: [root at localhost build]# /etc/init.d/freeswitch start Starting freeswitch: [ OK ] [root at localhost build]# /etc/init.d/freeswitch status freeswitch dead but subsys locked Did I miss somthing here? Please help. Thanks a lot. dynaguy Here is the copy of the origenal freeswitch.init.centos5 ---------------------------- [root at localhost ~]# cat /etc/init.d/freeswitch #!/bin/bash # # /etc/rc.d/init.d/freeswitch # # The FreeSwitch Open Source Voice Platform # # chkconfig: 345 89 14 # description: Starts and stops the freeswitch server daemon # processname: freeswitch # config: /usr/local/freeswitch/conf/freeswitch.conf # pidfile: /usr/local/freeswitch/log/freeswitch.pid # # Source function library. . /etc/init.d/functions PROG_NAME=freeswitch PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_USER=${FS_USER-freeswitch} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} LOCK_FILE=/var/lock/subsys/freeswitch FREESWITCH_ARGS="-nc" RETVAL=0 # Source options file if [ -f /etc/sysconfig/freeswitch ]; then . /etc/sysconfig/freeswitch fi # start() { echo -n "Starting $PROG_NAME: " if [ -e $LOCK_FILE ]; then if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then echo echo -n $"$PROG_NAME is already running."; failure $"$PROG_NAME is already running."; echo return 1 fi fi cd $FS_HOME daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" echo RETVAL=$? [ $RETVAL -eq 0 ] && touch $LOCK_FILE; echo return $RETVAL } stop() { echo -n "Shutting down $PROG_NAME: " if [ ! -e $LOCK_FILE ]; then echo echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running." failure $"cannot stop $PROG_NAME: $PROG_NAME is not running." echo return 1; fi cd $FS_HOME $FS_FILE -stop > /dev/null 2>&1 killproc $PROG_NAME RETVAL=$? echo [ $RETVAL -eq 0 ] && rm -f $LOCK_FILE; return $RETVAL } rhstatus() { status $PROG_NAME; } case "$1" in start) start ;; stop) stop ;; status) status $PROG_NAME RETVAL=$? ;; restart) stop start ;; reload) # ;; condrestart) [ -f $PID_FILE ] && restart || : ;; *) echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" exit 1 ;; esac exit $RETVAL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/112e9099/attachment-0002.html From dynaguy at gmail.com Thu Mar 5 13:09:16 2009 From: dynaguy at gmail.com (dynaguy) Date: Thu, 5 Mar 2009 13:09:16 -0800 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot Message-ID: <2D9EE07B35234097977448BE8EDCEDA9@dell200> Hello, I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh install) and followed the instruction on the wiki installed the new Freeswitch v1.0.3. After the installation I can start FS by issus command "/usr/local/freeswitch/bin/freeswitch". After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and chmod it to 755. Reboot the server and FS is not running. If I try the start up script: [root at localhost build]# /etc/init.d/freeswitch start Starting freeswitch: [ OK ] [root at localhost build]# /etc/init.d/freeswitch status freeswitch dead but subsys locked Did I miss somthing here? Please help. Thanks a lot. dynaguy Here is the copy of the origenal freeswitch.init.centos5 ---------------------------- [root at localhost ~]# cat /etc/init.d/freeswitch #!/bin/bash # # /etc/rc.d/init.d/freeswitch # # The FreeSwitch Open Source Voice Platform # # chkconfig: 345 89 14 # description: Starts and stops the freeswitch server daemon # processname: freeswitch # config: /usr/local/freeswitch/conf/freeswitch.conf # pidfile: /usr/local/freeswitch/log/freeswitch.pid # # Source function library. . /etc/init.d/functions PROG_NAME=freeswitch PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_USER=${FS_USER-freeswitch} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} LOCK_FILE=/var/lock/subsys/freeswitch FREESWITCH_ARGS="-nc" RETVAL=0 # Source options file if [ -f /etc/sysconfig/freeswitch ]; then . /etc/sysconfig/freeswitch fi # start() { echo -n "Starting $PROG_NAME: " if [ -e $LOCK_FILE ]; then if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then echo echo -n $"$PROG_NAME is already running."; failure $"$PROG_NAME is already running."; echo return 1 fi fi cd $FS_HOME daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" echo RETVAL=$? [ $RETVAL -eq 0 ] && touch $LOCK_FILE; echo return $RETVAL } stop() { echo -n "Shutting down $PROG_NAME: " if [ ! -e $LOCK_FILE ]; then echo echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running." failure $"cannot stop $PROG_NAME: $PROG_NAME is not running." echo return 1; fi cd $FS_HOME $FS_FILE -stop > /dev/null 2>&1 killproc $PROG_NAME RETVAL=$? echo [ $RETVAL -eq 0 ] && rm -f $LOCK_FILE; return $RETVAL } rhstatus() { status $PROG_NAME; } case "$1" in start) start ;; stop) stop ;; status) status $PROG_NAME RETVAL=$? ;; restart) stop start ;; reload) # ;; condrestart) [ -f $PID_FILE ] && restart || : ;; *) echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" exit 1 ;; esac exit $RETVAL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/a48ef230/attachment-0002.html From stevecrozz at gmail.com Thu Mar 5 14:26:27 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 5 Mar 2009 14:26:27 -0800 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: References: Message-ID: <11990ade0903051426h4390602fy6f92b9020618fba8@mail.gmail.com> Your script at /etc/init.d/freeswitch is probably not referenced anywhere in your init sequence. You should read some documentation on the boot process for your system, which is probably something like this: http://www.redhat.com/docs/manuals/linux/RHL-9-Manual/ref-guide/s1-boot-init-shutdown-sysv.html Someone else might have more specifics for you. --Stephen On Thu, Mar 5, 2009 at 12:19 PM, Dyna Guy wrote: > I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh > install) and followed the instruction on the wiki installed the new > Freeswitch v1.0.3.? After the installation I can start FS by issus command > "/usr/local/freeswitch/bin/freeswitch". > > After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and > chmod it to 755. Reboot the server and FS is not running. > > If I try the start up script: > [root at localhost build]# /etc/init.d/freeswitch start > Starting freeswitch:?????????????????????????????????????? [? OK? ] > [root at localhost build]# /etc/init.d/freeswitch status > freeswitch dead but subsys locked > Did I miss somthing here? Please help. Thanks a lot. > > dynaguy > > > > > > Here is the copy of the origenal freeswitch.init.centos5 > ---------------------------- > [root at localhost ~]# cat /etc/init.d/freeswitch > #!/bin/bash > # > #?????? /etc/rc.d/init.d/freeswitch > # > #?????? The FreeSwitch Open Source Voice Platform > # > #? chkconfig: 345 89 14 > #? description: Starts and stops the freeswitch server daemon > #? processname: freeswitch > #? config: /usr/local/freeswitch/conf/freeswitch.conf > #? pidfile: /usr/local/freeswitch/log/freeswitch.pid > # > # Source function library. > . /etc/init.d/functions > PROG_NAME=freeswitch > PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} > FS_USER=${FS_USER-freeswitch} > FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} > FS_HOME=${FS_HOME-/usr/local/freeswitch} > LOCK_FILE=/var/lock/subsys/freeswitch > FREESWITCH_ARGS="-nc" > RETVAL=0 > # Source options file > if [ -f /etc/sysconfig/freeswitch ]; then > ??????? . /etc/sysconfig/freeswitch > fi > # > start() { > ??????? echo -n "Starting $PROG_NAME: " > ??????? if [ -e $LOCK_FILE ]; then > ??????????? if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then > ??????????????? echo > ??????????????? echo -n $"$PROG_NAME is already running."; > ??????????????? failure $"$PROG_NAME is already running."; > ??????????????? echo > ??????????????? return 1 > ??????????? fi > ??????? fi > ??????? cd $FS_HOME > ??????? daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE > $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" > ??????????????? echo > ??????????????? RETVAL=$? > ??????? [ $RETVAL -eq 0 ] && touch $LOCK_FILE; > ??????? echo > ??????? return $RETVAL > } > stop() { > ??????? echo -n "Shutting down $PROG_NAME: " > ??????? if [ ! -e $LOCK_FILE ]; then > ??????????? echo > ??????????? echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running." > ??????????? failure $"cannot stop $PROG_NAME: $PROG_NAME is not running." > ??????????? echo > ??????????? return 1; > ??????? fi > ??????? cd $FS_HOME > ??????? $FS_FILE -stop > /dev/null 2>&1 > ??????? killproc $PROG_NAME > ??????? RETVAL=$? > ??????? echo > ??????? [ $RETVAL -eq 0 ] &&? rm -f $LOCK_FILE; > ??????? return $RETVAL > } > rhstatus() { > ??????? status $PROG_NAME; > } > case "$1" in > ??? start) > ??????? start > ??????? ;; > ??? stop) > ??????? stop > ??????? ;; > ??? status) > ??????? status $PROG_NAME > ??????? RETVAL=$? > ??????? ;; > ??? restart) > ??????? stop > ??????? start > ??????? ;; > ??? reload) > #??????? #??????? kill -HUP or by restarting the daemons, in a manner similar > #??????? to restart above> > ??????? ;; > ??? condrestart) > ??????? [ -f $PID_FILE ] && restart || : > ??????? ;; > ??? *) > ??????? echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" > ??????? exit 1 > ??????? ;; > esac > exit $RETVAL > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From e at musinghalfwit.org Thu Mar 5 14:38:42 2009 From: e at musinghalfwit.org (Eric Liedtke) Date: Thu, 5 Mar 2009 16:38:42 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) Message-ID: <20090305223842.GA31993@pointone.com> Greetings, I've been using FS in production on this rev (I realize it's pretty far behind current) and it's been running well, save 1 issue. The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I have 2 sip profiles created , 1 per ip interface. This is being used to terminate traffic to a provider so calls are only 1 direction. They come into the private side profile, get routed via dialplan to the gateway defined in the external profile and on to the vendor. Pretty simple. I have noticed that under load (50 or so cps with ~800-900 bridged calls up) that over time some channels on the public side seem to get "stuck". Due to the nature of how this is being used , I would expect both sip profiles to show the same number of channels in use any time i do a 'sofia status' ( or at least be within a channel or 2 of each other). However after a day of heavy use I had a disparity of ~250 channels. These extra channels also seem to put some continual load on the 'system cpu' as well , reported via top. Of course due to the load on the box I have to keep logging turned way down. So I've been trying to troubleshoot it as best I can. Last night I grabbed a core file and started in with GDB today. I found the 120 or so threads that represented real active calls when I took the corefile, I also found ~250 threads that appeared to be stuck in the CS_NEW state. The backtraces on all of them looks the same, annotated below. I walked through the code path by hand , based on the bt's and I don't see how this could be happening unless it's a locking issue. But as far as I can tell each session has it's own mutex defined in the switch_core_session_t struct, so I wouldn't think they would be stepping on each other. I also would have expected if it were something of a deadlock nature it would stop processing calls all together. I grabbed the commands from the .gdbinit (super handy btw!!) and have been trolling through the variables to try to ascertain something about why these threads seem to be stuck, but am not having much luck even coming up with a scenario to try to replicate the issue. If anyone has any pointers as to where I might look next it would be greatly appreciated. We will be updating to the newest release soon, however I was hoping to nail down what is going so I can systematically replicate it and verify by testing in the lab that it is fixed , rather than just pushing the new release to produvction and hoping. Thanks in advance for any tips/pointers anyone may have. -e ......bt and bt full for a single "hung" thread #0 0xb7fd5410 in __kernel_vsyscall () #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/switch_core_state_machine.c:462 #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, obj=0x95fe270) at src/switch_core_session.c:853 #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/thread.c:138 #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 (gdb) bt full #0 0xb7fd5410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 No locals. #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/switch_core_state_machine.c:462 exception = 0 '\0' state = endstate = CS_NEW endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xb73b1720 application_state_handler = thread_id = 3085554955 env = {{__jmpbuf = {134603552, -1428248680, -1461722504, 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, 9184, 1, 2976641592, 2833244792, 3086590960, 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, 3085458203, 3086590960, 2976606624, 134564192, 2833244904}}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, obj=0x95fe270) at src/switch_core_session.c:853 session = (switch_core_session_t *) 0x95fe270 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/thread.c:138 No locals. #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 From brian at freeswitch.org Thu Mar 5 14:52:43 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 16:52:43 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <20090305223842.GA31993@pointone.com> References: <20090305223842.GA31993@pointone.com> Message-ID: Well the rules usually state that you try SVN trunk then report a jira if the problem persists but since you're 2000+ revs behind chances are we already fixed this issue. Are you using bypass media? /b On Mar 5, 2009, at 4:38 PM, Eric Liedtke wrote: > Greetings, > > I've been using FS in production on this rev (I realize it's pretty > far > behind current) and it's been running well, save 1 issue. From mrene_lists at avgs.ca Thu Mar 5 14:55:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 5 Mar 2009 17:55:33 -0500 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <20090305223842.GA31993@pointone.com> References: <20090305223842.GA31993@pointone.com> Message-ID: <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> HI, If you suspect a bug, the place to report it is JIRA. See: http://wiki.freeswitch.org/wiki/Reporting_Bugs . This gives the whole team a way of following up on issues. Also can you upgrade to svn trunk? A lot of fixes gets committed daily, so its good to stay up to date. As you seem familiar with GDB, you may symlink the .gdbinit file in the support-d/ folder to your home directory. This will give you some FS-specific macros such as "list_sessions" which will dump a list of uuids to session pointers. In your jira, make sure you include "thread apply all bt", "list_sessions" and show channels (this one goes in FS) but PLEASE update to svn trunk and test again to see if it still happens. Also, are you using proxy/bypass media or just the default? Math On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote: > Greetings, > > I've been using FS in production on this rev (I realize it's pretty > far > behind current) and it's been running well, save 1 issue. > > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I > have > 2 sip profiles created , 1 per ip interface. This is being used to > terminate traffic to a provider so calls are only 1 direction. They > come > into the private side profile, get routed via dialplan to the gateway > defined in the external profile and on to the vendor. Pretty simple. > > I have noticed that under load (50 or so cps with ~800-900 bridged > calls up) > that over time some channels on the public side seem to get > "stuck". Due to > the nature of how this is being used , I would expect both sip > profiles to show > the same number of channels in use any time i do a 'sofia > status' ( or at least > be within a channel or 2 of each other). However after a day of > heavy use I had > a disparity of ~250 channels. These extra channels also seem to put > some > continual load on the 'system cpu' as well , reported via top. > > Of course due to the load on the box I have to keep logging turned way > down. So I've been trying to troubleshoot it as best I can. > > Last night I grabbed a core file and started in with GDB today. I > found > the 120 or so threads that represented real active calls when I took > the > corefile, I also found ~250 threads that appeared to be stuck in the > CS_NEW state. The backtraces on all of them looks the same, > annotated below. > > I walked through the code path by hand , based on the bt's and I > don't see how > this could be happening unless it's a locking issue. But as far as > I can tell > each session has it's own mutex defined in the > switch_core_session_t struct, > so I wouldn't think they would be stepping on each other. I also > would have expected > if it were something of a deadlock nature it would stop processing > calls all > together. > > I grabbed the commands from the .gdbinit (super handy btw!!) and > have been trolling > through the variables to try to ascertain something about why these > threads seem to > be stuck, but am not having much luck even coming up with a scenario > to try > to replicate the issue. > > If anyone has any pointers as to where I might look next it would be > greatly > appreciated. > > We will be updating to the newest release soon, however I was hoping > to nail down > what is going so I can systematically replicate it and verify by > testing in the lab > that it is fixed , rather than just pushing the new release to > produvction and hoping. > > Thanks in advance for any tips/pointers anyone may have. > > -e > > ......bt and bt full for a single "hung" thread > > > #0 0xb7fd5410 in __kernel_vsyscall () > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > switch_core_state_machine.c:462 > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > obj=0x95fe270) at src/switch_core_session.c:853 > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > thread.c:138 > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > (gdb) bt full > #0 0xb7fd5410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > No locals. > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > switch_core_state_machine.c:462 > exception = 0 '\0' > state = > endstate = CS_NEW > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xb73b1720 > application_state_handler = > thread_id = 3085554955 > env = {{__jmpbuf = {134603552, -1428248680, -1461722504, > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, > 9184, 1, 2976641592, 2833244792, 3086590960, > 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, > 3085458203, 3086590960, 2976606624, > 134564192, 2833244904}}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > obj=0x95fe270) at src/switch_core_session.c:853 > session = (switch_core_session_t *) 0x95fe270 > event = > event_str = 0x0 > val = > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > thread.c:138 > No locals. > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From e at musinghalfwit.org Thu Mar 5 15:19:00 2009 From: e at musinghalfwit.org (Eric Liedtke) Date: Thu, 5 Mar 2009 17:19:00 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: References: <20090305223842.GA31993@pointone.com> Message-ID: <20090305231900.GB31993@pointone.com> Yeah I know ;) I didn't open a bug because my rev was so far behind. I was just looking for any advice for where to poke next. Troubleshooting this has been a fantastic introduction to some of the inner workings of freeswitch so I was hoping to see it through and learn as I went. To answer your question no we are not using bypass media. -e It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 04:52:43PM -0600 , Brian West said: > Well the rules usually state that you try SVN trunk then report a jira > if the problem persists but since you're 2000+ revs behind chances are > we already fixed this issue. Are you using bypass media? > > /b > > On Mar 5, 2009, at 4:38 PM, Eric Liedtke wrote: > > > Greetings, > > > > I've been using FS in production on this rev (I realize it's pretty > > far > > behind current) and it's been running well, save 1 issue. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From e at musinghalfwit.org Thu Mar 5 15:22:37 2009 From: e at musinghalfwit.org (Eric Liedtke) Date: Thu, 5 Mar 2009 17:22:37 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> References: <20090305223842.GA31993@pointone.com> <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> Message-ID: <20090305232237.GC31993@pointone.com> Yup, as I mentioned to brian didn't want to clog jira with a bug that's been fixed or report against a rev 2k+ revs behind. I was trying to work through it as a learning exercise. And yeah I actually added a bunch of stuff to the list_sessions function to spit out a variety of associated variables for each session looking for a pattern somewhere to clue me into what might be happening. No proxy or bypass media here, just defaults. I will keep at it and once we update the production systems, if the problem persists I will open a bug in jira with all the neccessary goodies. Thanks -e It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM -0500 , Mathieu Rene said: > HI, > > If you suspect a bug, the place to report it is JIRA. See: http://wiki.freeswitch.org/wiki/Reporting_Bugs > . > This gives the whole team a way of following up on issues. > > Also can you upgrade to svn trunk? A lot of fixes gets committed > daily, so its good to stay up to date. > > As you seem familiar with GDB, you may symlink the .gdbinit file in > the support-d/ folder to your home directory. > This will give you some FS-specific macros such as "list_sessions" > which will dump a list of uuids to session pointers. > > In your jira, make sure you include "thread apply all bt", > "list_sessions" and show channels (this one goes in FS) but PLEASE > update to svn trunk and test again to see if it still happens. > > Also, are you using proxy/bypass media or just the default? > > Math > > On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote: > > > Greetings, > > > > I've been using FS in production on this rev (I realize it's pretty > > far > > behind current) and it's been running well, save 1 issue. > > > > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I > > have > > 2 sip profiles created , 1 per ip interface. This is being used to > > terminate traffic to a provider so calls are only 1 direction. They > > come > > into the private side profile, get routed via dialplan to the gateway > > defined in the external profile and on to the vendor. Pretty simple. > > > > I have noticed that under load (50 or so cps with ~800-900 bridged > > calls up) > > that over time some channels on the public side seem to get > > "stuck". Due to > > the nature of how this is being used , I would expect both sip > > profiles to show > > the same number of channels in use any time i do a 'sofia > > status' ( or at least > > be within a channel or 2 of each other). However after a day of > > heavy use I had > > a disparity of ~250 channels. These extra channels also seem to put > > some > > continual load on the 'system cpu' as well , reported via top. > > > > Of course due to the load on the box I have to keep logging turned way > > down. So I've been trying to troubleshoot it as best I can. > > > > Last night I grabbed a core file and started in with GDB today. I > > found > > the 120 or so threads that represented real active calls when I took > > the > > corefile, I also found ~250 threads that appeared to be stuck in the > > CS_NEW state. The backtraces on all of them looks the same, > > annotated below. > > > > I walked through the code path by hand , based on the bt's and I > > don't see how > > this could be happening unless it's a locking issue. But as far as > > I can tell > > each session has it's own mutex defined in the > > switch_core_session_t struct, > > so I wouldn't think they would be stepping on each other. I also > > would have expected > > if it were something of a deadlock nature it would stop processing > > calls all > > together. > > > > I grabbed the commands from the .gdbinit (super handy btw!!) and > > have been trolling > > through the variables to try to ascertain something about why these > > threads seem to > > be stuck, but am not having much luck even coming up with a scenario > > to try > > to replicate the issue. > > > > If anyone has any pointers as to where I might look next it would be > > greatly > > appreciated. > > > > We will be updating to the newest release soon, however I was hoping > > to nail down > > what is going so I can systematically replicate it and verify by > > testing in the lab > > that it is fixed , rather than just pushing the new release to > > produvction and hoping. > > > > Thanks in advance for any tips/pointers anyone may have. > > > > -e > > > > ......bt and bt full for a single "hung" thread > > > > > > #0 0xb7fd5410 in __kernel_vsyscall () > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > > switch_core_state_machine.c:462 > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > obj=0x95fe270) at src/switch_core_session.c:853 > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > > thread.c:138 > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > libpthread.so.0 > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > (gdb) bt full > > #0 0xb7fd5410 in __kernel_vsyscall () > > No symbol table info available. > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > No symbol table info available. > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > No symbol table info available. > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > No locals. > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > > switch_core_state_machine.c:462 > > exception = 0 '\0' > > state = > > endstate = CS_NEW > > endpoint_interface = > > driver_state_handler = (const switch_state_handler_table_t *) > > 0xb73b1720 > > application_state_handler = > > thread_id = 3085554955 > > env = {{__jmpbuf = {134603552, -1428248680, -1461722504, > > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, > > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, > > 9184, 1, 2976641592, 2833244792, 3086590960, > > 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, > > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, > > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, > > 3085458203, 3086590960, 2976606624, > > 134564192, 2833244904}}}} > > sig = > > __func__ = "switch_core_session_run" > > __PRETTY_FUNCTION__ = "switch_core_session_run" > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > obj=0x95fe270) at src/switch_core_session.c:853 > > session = (switch_core_session_t *) 0x95fe270 > > event = > > event_str = 0x0 > > val = > > __func__ = "switch_core_session_thread" > > __PRETTY_FUNCTION__ = "switch_core_session_thread" > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > > thread.c:138 > > No locals. > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > libpthread.so.0 > > No symbol table info available. > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raul at etellicom.com Thu Mar 5 15:32:52 2009 From: raul at etellicom.com (Raul Fragoso) Date: Thu, 05 Mar 2009 20:32:52 -0300 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: <2D9EE07B35234097977448BE8EDCEDA9@dell200> References: <2D9EE07B35234097977448BE8EDCEDA9@dell200> Message-ID: <1236295972.18566.38.camel@raul-laptop> Hi, and welcome to FreeSWITCH ! You've done everything right, now you only need to tell your system to run that init script during startup ;-) As root, do this: chkconfig --add freeswitch chkconfig --level 2345 freeswitch on That's all. Regards, Raul On Thu, 2009-03-05 at 13:09 -0800, dynaguy wrote: > Hello, > > I am a newbie to FS and I want learn it. So I setup a Centos 5.2 > (fresh install) and followed the instruction on the wiki installed the > new Freeswitch v1.0.3. After the installation I can start FS by issus > command "/usr/local/freeswitch/bin/freeswitch". > > > > After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch > and chmod it to 755. Reboot the server and FS is not running. > > > > If I try the start up script: > > [root at localhost build]# /etc/init.d/freeswitch start > Starting freeswitch: [ OK ] > > [root at localhost build]# /etc/init.d/freeswitch status > freeswitch dead but subsys locked > > Did I miss somthing here? Please help. Thanks a lot. > > > > dynaguy > > > > > > > > > > > > Here is the copy of the origenal freeswitch.init.centos5 > > ---------------------------- > > > [root at localhost ~]# cat /etc/init.d/freeswitch > #!/bin/bash > # > # /etc/rc.d/init.d/freeswitch > # > # The FreeSwitch Open Source Voice Platform > # > # chkconfig: 345 89 14 > # description: Starts and stops the freeswitch server daemon > # processname: freeswitch > # config: /usr/local/freeswitch/conf/freeswitch.conf > # pidfile: /usr/local/freeswitch/log/freeswitch.pid > # > > # Source function library. > . /etc/init.d/functions > > PROG_NAME=freeswitch > PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} > FS_USER=${FS_USER-freeswitch} > FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} > FS_HOME=${FS_HOME-/usr/local/freeswitch} > LOCK_FILE=/var/lock/subsys/freeswitch > FREESWITCH_ARGS="-nc" > RETVAL=0 > > # Source options file > if [ -f /etc/sysconfig/freeswitch ]; then > . /etc/sysconfig/freeswitch > fi > > # > > start() { > echo -n "Starting $PROG_NAME: " > if [ -e $LOCK_FILE ]; then > if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then > echo > echo -n $"$PROG_NAME is already running."; > failure $"$PROG_NAME is already running."; > echo > return 1 > fi > fi > cd $FS_HOME > daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE > $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" > echo > RETVAL=$? > [ $RETVAL -eq 0 ] && touch $LOCK_FILE; > echo > return $RETVAL > } > > stop() { > echo -n "Shutting down $PROG_NAME: " > if [ ! -e $LOCK_FILE ]; then > echo > echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not > running." > failure $"cannot stop $PROG_NAME: $PROG_NAME is not > running." > echo > return 1; > fi > cd $FS_HOME > $FS_FILE -stop > /dev/null 2>&1 > killproc $PROG_NAME > RETVAL=$? > echo > [ $RETVAL -eq 0 ] && rm -f $LOCK_FILE; > return $RETVAL > } > > rhstatus() { > status $PROG_NAME; > } > > case "$1" in > start) > start > ;; > stop) > stop > ;; > status) > status $PROG_NAME > RETVAL=$? > ;; > restart) > stop > start > ;; > reload) > # # kill -HUP or by restarting the daemons, in a manner similar > # to restart above> > ;; > condrestart) > [ -f $PID_FILE ] && restart || : > ;; > *) > echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" > exit 1 > ;; > esac > exit $RETVAL > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Mar 5 15:39:05 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 17:39:05 -0600 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: <1236295972.18566.38.camel@raul-laptop> References: <2D9EE07B35234097977448BE8EDCEDA9@dell200> <1236295972.18566.38.camel@raul-laptop> Message-ID: You might not wanna start it in level 2... network might not be up yet. /b On Mar 5, 2009, at 5:32 PM, Raul Fragoso wrote: > Hi, and welcome to FreeSWITCH ! > > You've done everything right, now you only need to tell your system to > run that init script during startup ;-) > As root, do this: > chkconfig --add freeswitch > chkconfig --level 2345 freeswitch on > > That's all. > > Regards, > > Raul From nik.middleton at noblesolutions.co.uk Thu Mar 5 15:39:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Mar 2009 23:39:45 -0000 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <20090305232237.GC31993@pointone.com> References: <20090305223842.GA31993@pointone.com><71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> <20090305232237.GC31993@pointone.com> Message-ID: Well if it's any consolation, I have a 4 day ish old copy of SVN and I have around 200 of these hung calls, though after an hour or so they did seem to clear. That said, FS made 138,330 call attempts today, not too shabby, and through out the call quality was as good as the first one. Not sure how to debug this one. Version: FreeSWITCH Version 1.0.trunk (12276) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric Liedtke Sent: 05 March 2009 23:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hung Channels (SVN Rev 10231) Yup, as I mentioned to brian didn't want to clog jira with a bug that's been fixed or report against a rev 2k+ revs behind. I was trying to work through it as a learning exercise. And yeah I actually added a bunch of stuff to the list_sessions function to spit out a variety of associated variables for each session looking for a pattern somewhere to clue me into what might be happening. No proxy or bypass media here, just defaults. I will keep at it and once we update the production systems, if the problem persists I will open a bug in jira with all the neccessary goodies. Thanks -e It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM -0500 , Mathieu Rene said: > HI, > > If you suspect a bug, the place to report it is JIRA. See: http://wiki.freeswitch.org/wiki/Reporting_Bugs > . > This gives the whole team a way of following up on issues. > > Also can you upgrade to svn trunk? A lot of fixes gets committed > daily, so its good to stay up to date. > > As you seem familiar with GDB, you may symlink the .gdbinit file in > the support-d/ folder to your home directory. > This will give you some FS-specific macros such as "list_sessions" > which will dump a list of uuids to session pointers. > > In your jira, make sure you include "thread apply all bt", > "list_sessions" and show channels (this one goes in FS) but PLEASE > update to svn trunk and test again to see if it still happens. > > Also, are you using proxy/bypass media or just the default? > > Math > > On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote: > > > Greetings, > > > > I've been using FS in production on this rev (I realize it's pretty > > far > > behind current) and it's been running well, save 1 issue. > > > > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I > > have > > 2 sip profiles created , 1 per ip interface. This is being used to > > terminate traffic to a provider so calls are only 1 direction. They > > come > > into the private side profile, get routed via dialplan to the gateway > > defined in the external profile and on to the vendor. Pretty simple. > > > > I have noticed that under load (50 or so cps with ~800-900 bridged > > calls up) > > that over time some channels on the public side seem to get > > "stuck". Due to > > the nature of how this is being used , I would expect both sip > > profiles to show > > the same number of channels in use any time i do a 'sofia > > status' ( or at least > > be within a channel or 2 of each other). However after a day of > > heavy use I had > > a disparity of ~250 channels. These extra channels also seem to put > > some > > continual load on the 'system cpu' as well , reported via top. > > > > Of course due to the load on the box I have to keep logging turned way > > down. So I've been trying to troubleshoot it as best I can. > > > > Last night I grabbed a core file and started in with GDB today. I > > found > > the 120 or so threads that represented real active calls when I took > > the > > corefile, I also found ~250 threads that appeared to be stuck in the > > CS_NEW state. The backtraces on all of them looks the same, > > annotated below. > > > > I walked through the code path by hand , based on the bt's and I > > don't see how > > this could be happening unless it's a locking issue. But as far as > > I can tell > > each session has it's own mutex defined in the > > switch_core_session_t struct, > > so I wouldn't think they would be stepping on each other. I also > > would have expected > > if it were something of a deadlock nature it would stop processing > > calls all > > together. > > > > I grabbed the commands from the .gdbinit (super handy btw!!) and > > have been trolling > > through the variables to try to ascertain something about why these > > threads seem to > > be stuck, but am not having much luck even coming up with a scenario > > to try > > to replicate the issue. > > > > If anyone has any pointers as to where I might look next it would be > > greatly > > appreciated. > > > > We will be updating to the newest release soon, however I was hoping > > to nail down > > what is going so I can systematically replicate it and verify by > > testing in the lab > > that it is fixed , rather than just pushing the new release to > > produvction and hoping. > > > > Thanks in advance for any tips/pointers anyone may have. > > > > -e > > > > ......bt and bt full for a single "hung" thread > > > > > > #0 0xb7fd5410 in __kernel_vsyscall () > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > > switch_core_state_machine.c:462 > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > obj=0x95fe270) at src/switch_core_session.c:853 > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > > thread.c:138 > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > libpthread.so.0 > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > (gdb) bt full > > #0 0xb7fd5410 in __kernel_vsyscall () > > No symbol table info available. > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > No symbol table info available. > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > No symbol table info available. > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > No locals. > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ > > switch_core_state_machine.c:462 > > exception = 0 '\0' > > state = > > endstate = CS_NEW > > endpoint_interface = > > driver_state_handler = (const switch_state_handler_table_t *) > > 0xb73b1720 > > application_state_handler = > > thread_id = 3085554955 > > env = {{__jmpbuf = {134603552, -1428248680, -1461722504, > > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, > > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, > > 9184, 1, 2976641592, 2833244792, 3086590960, > > 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, > > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, > > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, > > 3085458203, 3086590960, 2976606624, > > 134564192, 2833244904}}}} > > sig = > > __func__ = "switch_core_session_run" > > __PRETTY_FUNCTION__ = "switch_core_session_run" > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > obj=0x95fe270) at src/switch_core_session.c:853 > > session = (switch_core_session_t *) 0x95fe270 > > event = > > event_str = 0x0 > > val = > > __func__ = "switch_core_session_thread" > > __PRETTY_FUNCTION__ = "switch_core_session_thread" > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ > > thread.c:138 > > No locals. > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > libpthread.so.0 > > No symbol table info available. > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Thu Mar 5 15:43:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Mar 2009 17:43:35 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: References: <20090305223842.GA31993@pointone.com><71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> <20090305232237.GC31993@pointone.com> Message-ID: <063AB580-5ABD-44B8-9671-25FBE1A0483C@freeswitch.org> I would update... We fixed a few bugs related to hung calls in the past 24 hours. /b On Mar 5, 2009, at 5:39 PM, Nik Middleton wrote: > Well if it's any consolation, I have a 4 day ish old copy of SVN and I > have around 200 of these hung calls, though after an hour or so they > did > seem to clear. > > That said, FS made 138,330 call attempts today, not too shabby, and > through out the call quality was as good as the first one. Not sure > how > to debug this one. > > Version: FreeSWITCH Version 1.0.trunk (12276) From nik.middleton at noblesolutions.co.uk Thu Mar 5 15:59:44 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Mar 2009 23:59:44 -0000 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on Message-ID: Just curious here. I've always followed the fedora route but became disillusioned with the focus on the desktop rather than the server mode. Of late I've moved my servers to Centos. I felt the need for stable systems. Everyone seems to slate Centos, but to my surprise Anthony recommends Centos 5.2 which is nice to hear. Yes I know it's not bleeding edge, but I don't want that. Any reason why I should not be running Centos with FS? (I do plan on running 64 bit in future though) Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/ca34a2df/attachment-0002.html From mszlazak at aol.com Thu Mar 5 16:35:50 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 05 Mar 2009 19:35:50 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com><49AFBFA6.45B7.0079.0@averyschools.net><49AFC537.45B7.0079.0@averyschools.net><600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org><131e01c99db9$ed440ba0$c7cc22e0$@net> <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> Message-ID: <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> If as you say "people prefer forums" then that's the nature of the target market and controlling markets can be very difficult. So you go with the market to succeed. The "build it and they will come" attitude virtually never works well. -----Original Message----- From: Kristian Kielhofner To: freeswitch-users at lists.freeswitch.org Sent: Thu, 5 Mar 2009 12:39 pm Subject: Re: [Freeswitch-users] Please end the torment A bunch of telephony geeks and a 1900 number - what could go wrong? Anyways, I too don't understand why people prefer forums. I follow dozens of mailling lists and a half a dozen e-mail addresses without ever leaving my mail client. My mail client happens to be gmail, btw: - Much more customization, filtering, etc possible than any "web forum" - "Local" copies of all messages - Search is awesome, ever hear of Google? ;) Web forums are good when you have to serve ads to people to get paid. Other than that they are certainly not the ideal tool for the job. Besides (and don't take this as an insult) - have you ever compared the web forums to the mailing lists for projects that offer both? Say what you want to say about mailing lists and IRC but the reality (usually) is the l33tz all hang out here and web forums (almost always) end up with the same groups of n00bz circling around and around trying to figure out how to accomplish even the most basic of tasks. Obviously that can go both ways but as a rule of thumb the people that are usually in a position to help others typically prefer mailing lists (probably for some of the reasons I cited above). Or maybe they are just old gray hairs too stuck in their ways. I don't know. ;) On Thu, Mar 5, 2009 at 12:43 PM, Gregory Boehnlein wrote: > You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :) > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/a0baf878/attachment-0002.html From msc at freeswitch.org Thu Mar 5 16:52:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Mar 2009 16:52:36 -0800 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: References: Message-ID: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> > Everyone seems to slate Centos, but to my surprise Anthony recommends Centos > 5.2 which is nice to hear.? Yes I know it?s not bleeding edge, but I don?t > want that. Repeat the mantra: CentOS is boring and predictable; boring and predictable is perfect for real-time telephony systems. > Any reason why I should not be running Centos with FS? (I do plan on running > 64 bit in future though) None that I can think of unless you have a super cool Linux distro that none of us have ever heard of. 64 bit OS on 64 bit hardware is a good thing. :) -MC From jason at jasonjgw.net Thu Mar 5 17:01:24 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 6 Mar 2009 12:01:24 +1100 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> Message-ID: <20090306010124.GA12699@jdc.jasonjgw.net> mszlazak at aol.com wrote: > If as you say "people prefer forums" then that's the nature of the target > market and controlling markets can be very difficult. So you go with the > market to succeed. Most free software/open-source people I've encountered prefer mailing lists and don't like being forced to use a Web interface instead (unless it's the Web interface of their preferred Web mail provider, in which case they're not being compelled to use it). For some of us, a Web forum is hard and inconvenient to use, because it substitutes the forum operator's user interface for that of the user's preferred mail client. I have reasons for choosing the mail client that I use, and if I had to work via somebody else's Web interface instead it would probably result in my not participating at all. This list can also be accessed via the Web and over NNTP at gmane.org. For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user - just connect your news reader to news.gmane.org. You can also post from the newsgroup; the first time you do so, an automated e-mail message will arrive in your inbox requesting confirmation, for spam prevention purposes. I don't know whether it is possible to post from the gmane.org Web site. They use Xapian as their search tool, which, in my experience, usually places the most relevant posts near the top of the search results. From dynaguy at gmail.com Thu Mar 5 17:07:21 2009 From: dynaguy at gmail.com (Dyna Guy) Date: Thu, 5 Mar 2009 17:07:21 -0800 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: References: <2D9EE07B35234097977448BE8EDCEDA9@dell200> <1236295972.18566.38.camel@raul-laptop> Message-ID: Thanks for all your advices. I am still struggle to make FS start. I tried few things included: chkconfig --add freeswitch chkconfig --level 345 freeswitch on I also added a user "freeswitch" The problem is : if I run "/etc/init.d./freeswitch" manually, it says [OK] like this: [root at localhost build]# /etc/init.d/freeswitch start Starting freeswitch: [ OK ] But then I did a "ps aux | grep freeswitch" it doesn't show FS running. I am not a script guru. If I run FS from commandline as root: /usr/local/freeswitch/bin/freeswitch then I can see FS running. What did I missing here? dynaguy On Thu, Mar 5, 2009 at 3:39 PM, Brian West wrote: > You might not wanna start it in level 2... network might not be up yet. > > /b > > On Mar 5, 2009, at 5:32 PM, Raul Fragoso wrote: > > > Hi, and welcome to FreeSWITCH ! > > > > You've done everything right, now you only need to tell your system to > > run that init script during startup ;-) > > As root, do this: > > chkconfig --add freeswitch > > chkconfig --level 2345 freeswitch on > > > > That's all. > > > > Regards, > > > > Raul > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/2bd53811/attachment-0002.html From raul at etellicom.com Thu Mar 5 17:11:56 2009 From: raul at etellicom.com (Raul Fragoso) Date: Thu, 05 Mar 2009 22:11:56 -0300 Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: References: <2D9EE07B35234097977448BE8EDCEDA9@dell200> <1236295972.18566.38.camel@raul-laptop> Message-ID: <1236301916.18566.39.camel@raul-laptop> Ah yes, my mistake, thanks for the correction Brian. On Thu, 2009-03-05 at 17:39 -0600, Brian West wrote: > You might not wanna start it in level 2... network might not be up yet. > > /b > > On Mar 5, 2009, at 5:32 PM, Raul Fragoso wrote: > > > Hi, and welcome to FreeSWITCH ! > > > > You've done everything right, now you only need to tell your system to > > run that init script during startup ;-) > > As root, do this: > > chkconfig --add freeswitch > > chkconfig --level 2345 freeswitch on > > > > That's all. > > > > Regards, > > > > Raul > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From davidwdan at gmail.com Thu Mar 5 17:26:03 2009 From: davidwdan at gmail.com (David Dan) Date: Thu, 5 Mar 2009 20:26:03 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <20090306010124.GA12699@jdc.jasonjgw.net> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> <20090306010124.GA12699@jdc.jasonjgw.net> Message-ID: <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> Web forums are like the wild west of the internet. They offer nothing that a mailing list and good wiki can't handle. Just go take a look at the trixbox forums. The last thing you want it for someone that is looking into freeswitch for the first time, to come across something like this (The Beginning of the End for CE), 1 click off the front page. I'd really hate to see FS go down this Mob Rule path. On Thu, Mar 5, 2009 at 8:01 PM, Jason White wrote: > mszlazak at aol.com wrote: > > If as you say "people prefer forums" then that's the nature of the target > > market and controlling markets can be very difficult. So you go with the > > market to succeed. > > Most free software/open-source people I've encountered prefer mailing lists > and don't like being forced to use a Web interface instead (unless it's the > Web interface of their preferred Web mail provider, in which case they're > not > being compelled to use it). > > For some of us, a Web forum is hard and inconvenient to use, because it > substitutes the forum operator's user interface for that of the user's > preferred mail client. I have reasons for choosing the mail client that I > use, > and if I had to work via somebody else's Web interface instead it would > probably result in my not participating at all. > > This list can also be accessed via the Web and over NNTP at gmane.org. > > For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user > - > just connect your news reader to news.gmane.org. > > You can also post from the newsgroup; the first time you do so, an > automated > e-mail message will arrive in your inbox requesting confirmation, for spam > prevention purposes. > > I don't know whether it is possible to post from the gmane.org Web site. > They > use Xapian as their search tool, which, in my experience, usually places > the > most relevant posts near the top of the search results. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/7ab00aab/attachment-0002.html From dujinfang at gmail.com Thu Mar 5 18:00:05 2009 From: dujinfang at gmail.com (seven) Date: Fri, 6 Mar 2009 10:00:05 +0800 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49AFCD02.2000603@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> Message-ID: <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> try bridge ({effective_caller_id_name ="your_name",effective_caller_id_number="0000"}sofia/b-leg) On Mar 5, 2009, at 9:00 PM, rod wrote: > the A leg invite looks like this: > From: "Anonymous" > > it has been rewritten like this: > From: "Anonymous" > > rod > > rod wrote: >> Hi Brian, >> >> if I use the function effective_caller_id_number with my INVITE, I >> get this: >> >> From: "Anonymous" ;tag=17geyFjX5p0gS. >> >> this is not exactly what I'm looking for :p >> >> rod >> >> >> Brian West wrote: >> >>> Well this depends on how you're placing the call.. if its a standard >>> bridge you can on the A-Leg set >>> "effective_caller_id_number=000${caller_id_number}" before you call >>> bridge. >>> >>> Is the from already in the correct format? >>> >>> /b >>> >>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>> >>> >>>> Dear list, >>>> >>>> I'd like to rewrite the number in the Remote Party ID header and >>>> only in >>>> this header. >>>> >>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>> this >>>> example) in the RPID header : >>>> >>>> From: Anonymous;tag=1208367 >>>> Remote-Party-ID: >>>> >>> 123456 >>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>> >>>> should become: >>>> From: Anonymous;tag=1208367 >>>> Remote-Party-ID: >>>> >>> 000123456 >>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>> >>>> But the From field should remain unchanged. >>>> >>>> And how to strip this prefix: >>>> From: Anonymous;tag=1208367 >>>> Remote-Party-ID: >>>> >>> 000123456 >>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>> >>>> should become: >>>> From: Anonymous;tag=1208367 >>>> Remote-Party-ID: >>>> >>> 123456 >>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>> >>>> >>>> regards. >>>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Mar 5 19:59:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Mar 2009 21:59:34 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: References: <20090305223842.GA31993@pointone.com> <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca> <20090305232237.GC31993@pointone.com> Message-ID: <191c3a030903051959y4a89aafaw108f41648215b35e@mail.gmail.com> if they went away by themselves they must not have been hung? On Thu, Mar 5, 2009 at 5:39 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Well if it's any consolation, I have a 4 day ish old copy of SVN and I > have around 200 of these hung calls, though after an hour or so they did > seem to clear. > > That said, FS made 138,330 call attempts today, not too shabby, and > through out the call quality was as good as the first one. Not sure how > to debug this one. > > Version: FreeSWITCH Version 1.0.trunk (12276) > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric > Liedtke > Sent: 05 March 2009 23:23 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Hung Channels (SVN Rev 10231) > > Yup, as I mentioned to brian didn't want to clog jira with a bug that's > been fixed or report against a rev 2k+ revs behind. I was trying to work > through it as a learning exercise. And yeah I actually added a bunch of > stuff to the list_sessions function to spit out a variety of associated > variables for each session looking for a pattern somewhere to clue me > into what might be happening. > > No proxy or bypass media here, just defaults. > > I will keep at it and once we update the production systems, if the > problem persists I will open a bug in jira with all the neccessary > goodies. > > Thanks > -e > > It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM > -0500 , Mathieu Rene said: > > HI, > > > > If you suspect a bug, the place to report it is JIRA. See: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > . > > This gives the whole team a way of following up on issues. > > > > Also can you upgrade to svn trunk? A lot of fixes gets committed > > daily, so its good to stay up to date. > > > > As you seem familiar with GDB, you may symlink the .gdbinit file in > > the support-d/ folder to your home directory. > > This will give you some FS-specific macros such as "list_sessions" > > which will dump a list of uuids to session pointers. > > > > In your jira, make sure you include "thread apply all bt", > > "list_sessions" and show channels (this one goes in FS) but PLEASE > > update to svn trunk and test again to see if it still happens. > > > > Also, are you using proxy/bypass media or just the default? > > > > Math > > > > On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote: > > > > > Greetings, > > > > > > I've been using FS in production on this rev (I realize it's pretty > > > > far > > > behind current) and it's been running well, save 1 issue. > > > > > > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I > > > have > > > 2 sip profiles created , 1 per ip interface. This is being used to > > > terminate traffic to a provider so calls are only 1 direction. They > > > > come > > > into the private side profile, get routed via dialplan to the > gateway > > > defined in the external profile and on to the vendor. Pretty simple. > > > > > > I have noticed that under load (50 or so cps with ~800-900 bridged > > > calls up) > > > that over time some channels on the public side seem to get > > > "stuck". Due to > > > the nature of how this is being used , I would expect both sip > > > profiles to show > > > the same number of channels in use any time i do a 'sofia > > > status' ( or at least > > > be within a channel or 2 of each other). However after a day of > > > heavy use I had > > > a disparity of ~250 channels. These extra channels also seem to put > > > > some > > > continual load on the 'system cpu' as well , reported via top. > > > > > > Of course due to the load on the box I have to keep logging turned > way > > > down. So I've been trying to troubleshoot it as best I can. > > > > > > Last night I grabbed a core file and started in with GDB today. I > > > found > > > the 120 or so threads that represented real active calls when I took > > > > the > > > corefile, I also found ~250 threads that appeared to be stuck in the > > > CS_NEW state. The backtraces on all of them looks the same, > > > annotated below. > > > > > > I walked through the code path by hand , based on the bt's and I > > > don't see how > > > this could be happening unless it's a locking issue. But as far as > > > > I can tell > > > each session has it's own mutex defined in the > > > switch_core_session_t struct, > > > so I wouldn't think they would be stepping on each other. I also > > > would have expected > > > if it were something of a deadlock nature it would stop processing > > > calls all > > > together. > > > > > > I grabbed the commands from the .gdbinit (super handy btw!!) and > > > have been trolling > > > through the variables to try to ascertain something about why these > > > > threads seem to > > > be stuck, but am not having much luck even coming up with a scenario > > > > to try > > > to replicate the issue. > > > > > > If anyone has any pointers as to where I might look next it would be > > > > greatly > > > appreciated. > > > > > > We will be updating to the newest release soon, however I was hoping > > > > to nail down > > > what is going so I can systematically replicate it and verify by > > > testing in the lab > > > that it is fixed , rather than just pushing the new release to > > > produvction and hoping. > > > > > > Thanks in advance for any tips/pointers anyone may have. > > > > > > -e > > > > > > ......bt and bt full for a single "hung" thread > > > > > > > > > #0 0xb7fd5410 in __kernel_vsyscall () > > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at > src/ > > > switch_core_state_machine.c:462 > > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > > obj=0x95fe270) at src/switch_core_session.c:853 > > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at > threadproc/unix/ > > > thread.c:138 > > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > > libpthread.so.0 > > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > (gdb) bt full > > > #0 0xb7fd5410 in __kernel_vsyscall () > > > No symbol table info available. > > > #1 0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6 > > > No symbol table info available. > > > #2 0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6 > > > No symbol table info available. > > > #3 0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143 > > > No locals. > > > #4 0xb7e9da03 in switch_core_session_run (session=0x95fe270) at > src/ > > > switch_core_state_machine.c:462 > > > exception = 0 '\0' > > > state = > > > endstate = CS_NEW > > > endpoint_interface = > > > driver_state_handler = (const switch_state_handler_table_t *) > > > > 0xb73b1720 > > > application_state_handler = > > > thread_id = 3085554955 > > > env = {{__jmpbuf = {134603552, -1428248680, -1461722504, > > > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, > > > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, > > > 9184, 1, 2976641592, 2833244792, 3086590960, > > > 168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, > > > > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, > > > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, > > > > 3085458203, 3086590960, 2976606624, > > > 134564192, 2833244904}}}} > > > sig = > > > __func__ = "switch_core_session_run" > > > __PRETTY_FUNCTION__ = "switch_core_session_run" > > > #5 0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, > > > obj=0x95fe270) at src/switch_core_session.c:853 > > > session = (switch_core_session_t *) 0x95fe270 > > > event = > > > event_str = 0x0 > > > val = > > > __func__ = "switch_core_session_thread" > > > __PRETTY_FUNCTION__ = "switch_core_session_thread" > > > #6 0xb7efd916 in dummy_worker (opaque=0x9ada840) at > threadproc/unix/ > > > thread.c:138 > > > No locals. > > > #7 0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ > > > libpthread.so.0 > > > No symbol table info available. > > > #8 0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/3a3de389/attachment-0002.html From anthony.minessale at gmail.com Thu Mar 5 20:02:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Mar 2009 22:02:34 -0600 Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231) In-Reply-To: <20090305231900.GB31993@pointone.com> References: <20090305223842.GA31993@pointone.com> <20090305231900.GB31993@pointone.com> Message-ID: <191c3a030903052002s2db4e53bi94e489a376b0ad18@mail.gmail.com> in your case you will have no choice but to update. Please do a fresh checkout as the build system has also drastically changed. On Thu, Mar 5, 2009 at 5:19 PM, Eric Liedtke wrote: > Yeah I know ;) I didn't open a bug because my rev was so far behind. I > was just looking for any advice for where to poke next. Troubleshooting > this has been a fantastic introduction to some of the inner workings of > freeswitch so I was hoping to see it through and learn as I went. > > To answer your question no we are not using bypass media. > > -e > > It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 04:52:43PM -0600 , > Brian West said: > > Well the rules usually state that you try SVN trunk then report a jira > > if the problem persists but since you're 2000+ revs behind chances are > > we already fixed this issue. Are you using bypass media? > > > > /b > > > > On Mar 5, 2009, at 4:38 PM, Eric Liedtke wrote: > > > > > Greetings, > > > > > > I've been using FS in production on this rev (I realize it's pretty > > > far > > > behind current) and it's been running well, save 1 issue. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090305/f9c2b14c/attachment-0002.html From mszlazak at aol.com Thu Mar 5 22:39:50 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 06 Mar 2009 01:39:50 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com><8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com><20090306010124.GA12699@jdc.jasonjgw.net> <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> Message-ID: <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> Again, if your target market prefers lists, then go with list. If they prefer forums then it's forums. The point is that it's not about what a few like, it's about the mob but the right mob. -----Original Message----- From: David Dan To: freeswitch-users at lists.freeswitch.org Sent: Thu, 5 Mar 2009 5:26 pm Subject: Re: [Freeswitch-users] Please end the torment Web forums are like the wild west of the internet.? They offer nothing that a mailing list and good wiki can't handle. Just go take a look at the trixbox forums. The last thing you want it for someone that is looking into freeswitch for the first time, to come across something like this (The Beginning of the End for CE), 1 click off the front page.? I'd really hate to see FS go down this Mob Rule path. On Thu, Mar 5, 2009 at 8:01 PM, Jason White wrote: mszlazak at aol.com wrote: > If as you say "people prefer forums" then that's the nature of the target > market and controlling markets can be very difficult. So you go with the > market to succeed. Most free software/open-source people I've encountered prefer mailing lists and don't like being forced to use a Web interface instead (unless it's the Web interface of their preferred Web mail provider, in which case they're not being compelled to use it). For some of us, a Web forum is hard and inconvenient to use, because it substitutes the forum operator's user interface for that of the user's preferred mail client. I have reasons for choosing the mail client that I use, and if I had to work via somebody else's Web interface instead it would probably result in my not participating at all. This list can also be accessed via the Web and over NNTP at gmane.org. For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user - just connect your news reader to news.gmane.org. You can also post from the newsgroup; the first time you do so, an automated e-mail message will arrive in your inbox requesting confirmation, for spam prevention purposes. I don't know whether it is possible to post from the gmane.org Web site. They use Xapian as their search tool, which, in my experience, usually places the most relevant posts near the top of the search results. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/a29d0248/attachment-0002.html From mashudiflexi at telkom.co.id Thu Mar 5 23:02:45 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Fri, 06 Mar 2009 14:02:45 +0700 Subject: [Freeswitch-users] make freeswitch-snapshot Message-ID: <49B0CA95.5080101@telkom.co.id> Hi Folk, i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86 here is the error : /usr/bin/ld: cannot find -lodbc collect2: ld returned 1 exit status make[2]: *** [libfreeswitch.la] Error 1 Making all in src Making all in mod making all mod_amr make[5]: *** No rule to make target `/usr/src/freeswitch-snapshot/libfreeswitch.la', needed by `mod_amr.so'. Stop. make[4]: *** [all] Error 1 make[3]: *** [mod_amr-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Did I miss something ? thank you for your support. mashudi ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From jason at jasonjgw.net Thu Mar 5 23:00:36 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 6 Mar 2009 18:00:36 +1100 Subject: [Freeswitch-users] make freeswitch-snapshot In-Reply-To: <49B0CA95.5080101@telkom.co.id> References: <49B0CA95.5080101@telkom.co.id> Message-ID: <20090306070036.GA24314@jdc.jasonjgw.net> mashudi wrote: > i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86 > here is the error : > > /usr/bin/ld: cannot find -lodbc Have you installed the ODBC library and its development headers? Are they the latest version? It's failing to find the ODBC library. From stevecrozz at gmail.com Thu Mar 5 23:03:51 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 5 Mar 2009 23:03:51 -0800 Subject: [Freeswitch-users] make freeswitch-snapshot In-Reply-To: <20090306070036.GA24314@jdc.jasonjgw.net> References: <49B0CA95.5080101@telkom.co.id> <20090306070036.GA24314@jdc.jasonjgw.net> Message-ID: <11990ade0903052303s21726d9qa956c52d52418585@mail.gmail.com> I think you need to install the debian package 'unixodbc-dev' --Stephen On Thu, Mar 5, 2009 at 11:00 PM, Jason White wrote: > mashudi wrote: >> i got error while conduct ?./make ?freeswitch-snapshot on debian 2.6 x86 >> here is the error : >> >> /usr/bin/ld: cannot find -lodbc > > Have you installed the ODBC library and its development headers? Are they the > latest version? > > It's failing to find the ODBC library. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kawarod at laposte.net Thu Mar 5 23:51:31 2009 From: kawarod at laposte.net (rod) Date: Fri, 06 Mar 2009 11:51:31 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> Message-ID: <49B0D603.502@laposte.net> using these functions like this did nothing on the SIP INVITE packet :'( seven wrote: > try > bridge > ({effective_caller_id_name > ="your_name",effective_caller_id_number="0000"}sofia/b-leg) > > On Mar 5, 2009, at 9:00 PM, rod wrote: > > >> the A leg invite looks like this: >> From: "Anonymous" >> >> it has been rewritten like this: >> From: "Anonymous" >> >> rod >> >> rod wrote: >> >>> Hi Brian, >>> >>> if I use the function effective_caller_id_number with my INVITE, I >>> get this: >>> >>> From: "Anonymous" ;tag=17geyFjX5p0gS. >>> >>> this is not exactly what I'm looking for :p >>> >>> rod >>> >>> >>> Brian West wrote: >>> >>> >>>> Well this depends on how you're placing the call.. if its a standard >>>> bridge you can on the A-Leg set >>>> "effective_caller_id_number=000${caller_id_number}" before you call >>>> bridge. >>>> >>>> Is the from already in the correct format? >>>> >>>> /b >>>> >>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>> >>>> >>>> >>>>> Dear list, >>>>> >>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>> only in >>>>> this header. >>>>> >>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>> this >>>>> example) in the RPID header : >>>>> >>>>> From: Anonymous;tag=1208367 >>>>> Remote-Party-ID: >>>>> >>>> 123456 >>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>> >>>>> should become: >>>>> From: Anonymous;tag=1208367 >>>>> Remote-Party-ID: >>>>> >>>> 000123456 >>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>> >>>>> But the From field should remain unchanged. >>>>> >>>>> And how to strip this prefix: >>>>> From: Anonymous;tag=1208367 >>>>> Remote-Party-ID: >>>>> >>>> 000123456 >>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>> >>>>> should become: >>>>> From: Anonymous;tag=1208367 >>>>> Remote-Party-ID: >>>>> >>>> 123456 >>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>> >>>>> >>>>> regards. >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From mashudiflexi at telkom.co.id Fri Mar 6 00:18:01 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Fri, 06 Mar 2009 15:18:01 +0700 Subject: [Freeswitch-users] make freeswitch-snapshot In-Reply-To: <11990ade0903052303s21726d9qa956c52d52418585@mail.gmail.com> References: <49B0CA95.5080101@telkom.co.id> <20090306070036.GA24314@jdc.jasonjgw.net> <11990ade0903052303s21726d9qa956c52d52418585@mail.gmail.com> Message-ID: <49B0DC39.1070203@telkom.co.id> Yes, it works , I would like to say thank you to Stephen Crosby & Jason White. Stephen Crosby wrote: > I think you need to install the debian package 'unixodbc-dev' > > --Stephen > > On Thu, Mar 5, 2009 at 11:00 PM, Jason White wrote: > >> mashudi wrote: >> >>> i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86 >>> here is the error : >>> >>> /usr/bin/ld: cannot find -lodbc >>> >> Have you installed the ODBC library and its development headers? Are they the >> latest version? >> >> It's failing to find the ODBC library. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From gmaruzz at celliax.org Fri Mar 6 01:39:30 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 6 Mar 2009 10:39:30 +0100 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> Message-ID: <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> On Fri, Mar 6, 2009 at 1:52 AM, Michael Collins wrote: >> Everyone seems to slate Centos, but to my surprise Anthony recommends Centos >> 5.2 which is nice to hear.? Yes I know it?s not bleeding edge, but I don?t >> want that. > > Repeat the mantra: CentOS is boring and predictable; boring and > predictable is perfect for real-time telephony systems. > >> Any reason why I should not be running Centos with FS? (I do plan on running >> 64 bit in future though) > > None that I can think of unless you have a super cool Linux distro > that none of us have ever heard of. > Maybe, but just maybe, on CentOS you can have a problem running skypiax (the skype endpoint/trunk): after a couple days of inactivity the snd-dummy ALSA driver of CentOS (at least on 32 bit) seems to go into ininterruptable sleep, causing the Skype clients to go into that state (the state seen as "D" in top). But I'm not sure about this, maybe will not be confirmed, needs more investigation. The Jira I filed for this is: http://jira.freeswitch.org/browse/MODSKYPIAX-27 I had very good overall experiences with Ubuntu 8.04 LTS Hardy, and CentOS 5.2. BTW: since roughly one month, when the sqlite assert was fixed, the build on Windows Vista seems rock solid to me. From nik.middleton at noblesolutions.co.uk Fri Mar 6 03:04:25 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Mar 2009 11:04:25 -0000 Subject: [Freeswitch-users] Setting External IP Message-ID: Hi Guys, In External.xml in sip profiles I have Can I override these for a given gateway profile? I have one gateway that's expecting a local routed IP address due to the way that it's routed, but the other one expects the public IP, hence the need to make it gateway specific Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/724761bf/attachment-0002.html From Claudio.Cavalera at italtel.it Fri Mar 6 03:21:41 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Fri, 6 Mar 2009 12:21:41 +0100 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > Hello list, > I'm trying to track down a seg fault issue with a fs Revision: 11489 > Here is the backtrace pastebin: > http://pastebin.freeswitch.org/7009 > > but before digging the dump I would like to understand: am I the only > one having error like this in fs console: > "Error in my_thread_global_end(): 16 threads didn't exit" > > I'm asking this because googling around did not take me to > much relation > between this error and fs. > In fact as you can see the error does not have the usual fs logging > format with date time and logging level, it's just a yellow > line printed > out in console. Hello, I'm trying to track down the source of this "problem". For this reason I would like to redirect this message to a log file so that it could be compared and correlated with other logs. I'm starting fs with this command in a script: bin/freeswitch -nc -core -log /var/log/freeswitch -conf /usr/local/freeswitch/conf -db /usr/local/freeswitch/db >> /var/log/freeswitch/fs_redirection.log 2>> /var/log/freeswitch/fs_redirection.log do you think I'm safe and I will capture the error message or the -nc option could change the behaviour? Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From saeedahmad1981 at gmail.com Fri Mar 6 03:31:35 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Fri, 6 Mar 2009 12:31:35 +0100 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com><8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com><20090306010124.GA12699@jdc.jasonjgw.net><65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> Message-ID: <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> We need a poll. a) List b) Forum > (b) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/6565422d/attachment-0002.html From helmut.kuper at ewetel.de Fri Mar 6 03:31:16 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 06 Mar 2009 12:31:16 +0100 Subject: [Freeswitch-users] Setting External IP In-Reply-To: References: Message-ID: <49B10984.8080807@ewetel.de> Hi Nik, yes you can! regards Helmut On 06.03.2009 12:04, Nik Middleton wrote: > > Hi Guys, > > > > In External.xml in sip profiles I have > > > > > > > > > > Can I override these for a given gateway profile? I have one gateway > that?s expecting a local routed IP address due to the way that it?s > routed, but the other one expects the public IP, hence the need to > make it gateway specific > > > > Regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/814b6c09/attachment-0002.html From dujinfang at gmail.com Fri Mar 6 04:34:10 2009 From: dujinfang at gmail.com (dujinfang) Date: Fri, 6 Mar 2009 20:34:10 +0800 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B0D603.502@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> Message-ID: <46348D07-D227-42A5-A25D-A047CE5B1B63@gmail.com> How about this? bridge ({origination_caller_id_name ="your_name",origination_caller_id_number="0000"}sofia/b-leg) On Mar 6, 2009, at 3:51 PM, rod wrote: > using these functions like this did nothing on the SIP INVITE > packet :'( > > seven wrote: >> try >> bridge >> ({effective_caller_id_name >> ="your_name",effective_caller_id_number="0000"}sofia/b-leg) >> >> On Mar 5, 2009, at 9:00 PM, rod wrote: >> >> >>> the A leg invite looks like this: >>> From: "Anonymous" >>> >>> it has been rewritten like this: >>> From: "Anonymous" >>> >>> rod >>> >>> rod wrote: >>> >>>> Hi Brian, >>>> >>>> if I use the function effective_caller_id_number with my INVITE, I >>>> get this: >>>> >>>> From: "Anonymous" >>> 000000anonymous at 172.29.0.5>;tag=17geyFjX5p0gS. >>>> >>>> this is not exactly what I'm looking for :p >>>> >>>> rod >>>> >>>> >>>> Brian West wrote: >>>> >>>> >>>>> Well this depends on how you're placing the call.. if its a >>>>> standard >>>>> bridge you can on the A-Leg set >>>>> "effective_caller_id_number=000${caller_id_number}" before you >>>>> call >>>>> bridge. >>>>> >>>>> Is the from already in the correct format? >>>>> >>>>> /b >>>>> >>>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>>> >>>>> >>>>> >>>>>> Dear list, >>>>>> >>>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>>> only in >>>>>> this header. >>>>>> >>>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>>> this >>>>>> example) in the RPID header : >>>>>> >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 123456 >>>>>> @10.10.10.10 >>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> should become: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 000123456 >>>>>> @10.10.10.10 >>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> But the From field should remain unchanged. >>>>>> >>>>>> And how to strip this prefix: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 000123456 >>>>>> @10.10.10.10 >>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> should become: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 123456 >>>>>> @10.10.10.10 >>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> >>>>>> regards. >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pablosaro at gmail.com Fri Mar 6 04:54:42 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 10:54:42 -0200 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> Message-ID: <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> Hi guys, I'm using Bluewhite64 for my Linux Servers. No problems compiling and using FS, but not funny if you have dependencies problems (no yum or aptitude available to solve your problems). Really stable and secure Linux 64 bit distribution. Obviously, as the system administrator, you have to take care of keeping the system up to date with security patches and proper configurations. Regards Pablo On Fri, Mar 6, 2009 at 7:39 AM, Giovanni Maruzzelli wrote: > On Fri, Mar 6, 2009 at 1:52 AM, Michael Collins wrote: >>> Everyone seems to slate Centos, but to my surprise Anthony recommends Centos >>> 5.2 which is nice to hear.? Yes I know it?s not bleeding edge, but I don?t >>> want that. >> >> Repeat the mantra: CentOS is boring and predictable; boring and >> predictable is perfect for real-time telephony systems. >> >>> Any reason why I should not be running Centos with FS? (I do plan on running >>> 64 bit in future though) >> >> None that I can think of unless you have a super cool Linux distro >> that none of us have ever heard of. >> > > Maybe, but just maybe, on CentOS you can have a problem running > skypiax (the skype endpoint/trunk): after a couple days of inactivity > the snd-dummy ALSA driver of CentOS (at least on 32 bit) seems to go > into ininterruptable sleep, causing the Skype clients to go into that > state (the state seen as "D" in top). But I'm not sure about this, > maybe will not be confirmed, needs more investigation. The Jira I > filed for this is: http://jira.freeswitch.org/browse/MODSKYPIAX-27 > > I had very good overall experiences with Ubuntu 8.04 LTS Hardy, and > CentOS 5.2. BTW: since roughly one month, when the sqlite assert was > fixed, the build on Windows Vista seems rock solid to me. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From edpimentl at gmail.com Fri Mar 6 05:01:19 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 6 Mar 2009 08:01:19 -0500 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> Message-ID: <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> Anyone using uBuntu 8.10 and XEN? What has been your most stable VM / FS platform? Thanks in advance, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/1fa1bf9f/attachment-0002.html From sergey.kirillov at gmail.com Fri Mar 6 05:03:08 2009 From: sergey.kirillov at gmail.com (Sergey Kirillov) Date: Fri, 06 Mar 2009 15:03:08 +0200 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card Message-ID: <49B11F0C.6040706@gmail.com> Hi everybody, I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work. Can somebody help to to configure it? Problem log (Incoming call): 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel OpenZAP/1:1/2360012 [7473c92a-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->2360012 in context default 2009-03-06 14:58:14 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer OpenZAP/1:1/2360012 to XML[1000 at default] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->1000 in context default 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/opt/freeswitch/recordings/80503820933.2009-03-06-14-58-14.wav 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:1000 at 192.168.122.1:5061;transport=udp [748a2ba2-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:1/2360012! 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:15 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:26 [NOTICE] Span:0 Q.921() I frame in invalid state ignored Here are my config files --- openzap.conf -- [span wanpipe BRI_1] name => BRI_1 trunk_type => bri b-channel => 1:1-2 d-channel => 1:3 --- openzap.conf.xml --- --- wanpipe1.conf --- [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] wp1aft1 = wanpipe1, auto, API, Comment wp1aft2 = wanpipe1, auto, API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 5 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_HW_DTMF = NO [wp1aft1] HDLC_STREAMING = NO ACTIVE_CH = 1-15.17-31 IDLE_FLAG = 0x7E MTU = 240 MRU = 240 DATA_MUX = NO TDMV_HWEC = NO [wp1aft2] HDLC_STREAMING = YES ACTIVE_CH = 16 MTU = 1500 MRU = 1500 DATA_MUX = NO TDMV_HWEC = NO From pablosaro at gmail.com Fri Mar 6 05:41:01 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 11:41:01 -0200 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> Message-ID: <247f8100903060541v10ab6605hc1d4e4de52f3db9b@mail.gmail.com> I've tried FS on ESX, but not cheap. Works good for small environments. I've never virtualized FS for big or critical systems. Pablo On Fri, Mar 6, 2009 at 11:01 AM, EdPimentl wrote: > Anyone using uBuntu 8.10 and XEN? > What has been your most stable VM / FS platform? > Thanks in advance, > -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Mar 6 06:00:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Mar 2009 08:00:28 -0600 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: References: Message-ID: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> We are considering asking CentOS to make a "FS cut" set of packages ideal for a telephony server with one install choice. On Thu, Mar 5, 2009 at 5:59 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Just curious here. > > > > I?ve always followed the fedora route but became disillusioned with the > focus on the desktop rather than the server mode. Of late I?ve moved my > servers to Centos. I felt the need for stable systems. > > > > Everyone seems to slate Centos, but to my surprise Anthony recommends > Centos 5.2 which is nice to hear. Yes I know it?s not bleeding edge, but I > don?t want that. > > > > Any reason why I should not be running Centos with FS? (I do plan on > running 64 bit in future though) > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/5e35b2f4/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 6 06:21:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Mar 2009 08:21:08 -0600 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: References: Message-ID: <191c3a030903060621r2592a89i208a7cc9cf655d91@mail.gmail.com> I recall someone asking about this before and it has to be one of the depends. Do you load anything that is not enabled by default in the standard install. It says my_ in it could it be mysql ? Let's google "my_thread_global_end()" and see... Checking http://www.google.com/search?q=my_thread_global_end() hey! http://forums.mysql.com/read.php?10,153077,153077 Not sure if its the same thing or not but it looks pretty similar. On Fri, Mar 6, 2009 at 5:21 AM, Cavalera Claudio Luigi < Claudio.Cavalera at italtel.it> wrote: > freeswitch-users-bounces at lists.freeswitch.org wrote: > > Hello list, > > I'm trying to track down a seg fault issue with a fs Revision: 11489 > > Here is the backtrace pastebin: > > http://pastebin.freeswitch.org/7009 > > > > but before digging the dump I would like to understand: am I the only > > one having error like this in fs console: > > "Error in my_thread_global_end(): 16 threads didn't exit" > > > > I'm asking this because googling around did not take me to > > much relation > > between this error and fs. > > In fact as you can see the error does not have the usual fs logging > > format with date time and logging level, it's just a yellow > > line printed > > out in console. > > > Hello, > I'm trying to track down the source of this "problem". > For this reason I would like to redirect this message to a log file so > that it could be compared and correlated with other logs. > I'm starting fs with this command in a script: > bin/freeswitch -nc -core -log /var/log/freeswitch -conf > /usr/local/freeswitch/conf -db /usr/local/freeswitch/db >> > /var/log/freeswitch/fs_redirection.log 2>> > /var/log/freeswitch/fs_redirection.log > > do you think I'm safe and I will capture the error message or the -nc > option could change the behaviour? > > Thanks, > Claudio > > > 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-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/1956e7e9/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 6 06:23:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Mar 2009 08:23:23 -0600 Subject: [Freeswitch-users] Setting External IP In-Reply-To: References: Message-ID: <191c3a030903060623p16dabe4bw5f4ed1fb2d196ca4@mail.gmail.com> gateways are children of profiles so if you need them to be separate you need to make 2 profiles and run the other one on another IP or another port. On Fri, Mar 6, 2009 at 5:04 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > In External.xml in sip profiles I have > > > > > > > > > > Can I override these for a given gateway profile? I have one gateway > that?s expecting a local routed IP address due to the way that it?s routed, > but the other one expects the public IP, hence the need to make it gateway > specific > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/faedf871/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 6 06:27:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Mar 2009 08:27:27 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> <20090306010124.GA12699@jdc.jasonjgw.net> <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> Message-ID: <191c3a030903060627y2102ebccscadf0550f7ba0d57@mail.gmail.com> Did anybody notice my email from yesterday that shows how there already is a forum on voip-info that is linked to our homepage and nobody uses it? We can't take this poll until we have a list of volunteers who would manage any new online resources. On Fri, Mar 6, 2009 at 5:31 AM, Saeed Ahmed wrote: > We need a poll. > > a) List > b) Forum > > > (b) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/0764d3c7/attachment-0002.html From msc at freeswitch.org Fri Mar 6 08:12:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Mar 2009 08:12:37 -0800 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <191c3a030903060627y2102ebccscadf0550f7ba0d57@mail.gmail.com> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> <20090306010124.GA12699@jdc.jasonjgw.net> <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> <191c3a030903060627y2102ebccscadf0550f7ba0d57@mail.gmail.com> Message-ID: <87f2f3b90903060812s13bccf49ma7b31a1dadf5f272@mail.gmail.com> > Did anybody notice my email from yesterday that shows how there already is a > forum on voip-info that is linked to our homepage and nobody uses it? > > We can't take this poll until we have a list of volunteers who would manage > any new > online resources. Let's "end the torment of this thread" by giving an official request for the community: All of those who are willing and able to moderate a forum please stand up, go log in to voip-info.org and see what needs to be done to make it a usable forum. If you are willing to commit to managing the forum please email me at msc at freeswitch.org. There will *NOT* be a forum if we don't get at least one motivated community volunteer to take care of it. Thanks, MC From dujinfang at gmail.com Fri Mar 6 08:22:41 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 7 Mar 2009 00:22:41 +0800 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> References: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com> <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com> <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com> <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com> Message-ID: We are using ubuntu 8.04 in Xen(also hosted by ubuntu 8.04, Ubuntu 8.10 is not xen friendly) as our testing server. It works well, however we only use that to test our business logic, not press test at all. On Mar 6, 2009, at 9:01 PM, EdPimentl wrote: > Anyone using uBuntu 8.10 and XEN? > What has been your most stable VM / FS platform? > Thanks in advance, > -E > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From codecomplete at free.fr Fri Mar 6 08:30:30 2009 From: codecomplete at free.fr (Fred) Date: Fri, 06 Mar 2009 17:30:30 +0100 Subject: [Freeswitch-users] Lunchbox-type PC as small server? Message-ID: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Hello I'm looking for a small, lunchbox-like PC to build a small-form factor CRM server to sell to small companies. Ideally, it should have one PCI slot so that I can stick a voice card to connect to an analog phone line and run FreeSwitch as well. I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it doesn't have room for a PCI slot, and I'm concerned about its performance. I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit pricey, and might also not be fast enough to act as a server. Are there brands/models you think I should look at? Thank you. From vikas.sharma711 at gmail.com Thu Mar 5 22:55:43 2009 From: vikas.sharma711 at gmail.com (Vikas Sharma) Date: Fri, 6 Mar 2009 12:25:43 +0530 Subject: [Freeswitch-users] About FreeSwitch Message-ID: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> HI all, I am new to this. Can anybody tell me that freeSwitch can be used as PBX indecently? Can it be integrated with other pbx as a media server? If yes, what features it has as a media server? Thnax for any help. -- vikas sharma -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/bf0aa0ce/attachment-0002.html From rupa at rupa.com Fri Mar 6 03:36:59 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 6 Mar 2009 05:36:59 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com> <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com> <20090306010124.GA12699@jdc.jasonjgw.net> <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com> <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com> <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop> Message-ID: a) List On Fri, Mar 6, 2009 at 5:31 AM, Saeed Ahmed wrote: > We need a poll. > > a) List > b) Forum > > > (b) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/d76e47c7/attachment-0002.html From chris at cloudtel.com Thu Mar 5 22:54:23 2009 From: chris at cloudtel.com (Chris Burns) Date: Thu, 5 Mar 2009 22:54:23 -0800 Subject: [Freeswitch-users] make freeswitch-snapshot In-Reply-To: <49B0CA95.5080101@telkom.co.id> References: <49B0CA95.5080101@telkom.co.id> Message-ID: <200903052254.23746.chris@cloudtel.com> apt-get install unixodbc-dev On March 5, 2009 11:02:45 pm mashudi wrote: > Hi Folk, > i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86 > here is the error : > > /usr/bin/ld: cannot find -lodbc > collect2: ld returned 1 exit status > make[2]: *** [libfreeswitch.la] Error 1 > Making all in src > Making all in mod > > making all mod_amr > make[5]: *** No rule to make target > `/usr/src/freeswitch-snapshot/libfreeswitch.la', needed by > `mod_amr.so'. Stop. > make[4]: *** [all] Error 1 > make[3]: *** [mod_amr-all] Error 1 > make[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + make install + > +----------------------------------------------+ > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Did I miss something ? > thank you for your support. > > mashudi > > > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From contactramu at gmail.com Fri Mar 6 00:51:46 2009 From: contactramu at gmail.com (Ramu) Date: Fri, 6 Mar 2009 03:51:46 -0500 Subject: [Freeswitch-users] Freeswitch and Kamailio (OpenSer) Integration Message-ID: <439e75680903060051y5021b292l76286dfe927d0337@mail.gmail.com> Hi All, I would like to setup freswitch and kamailio as follows: Kamailio acts as Proxy and Registrator Freeswitch acts as a SBC and MediaServer (voicemail) Users will be reigstered to Kamailio Kamailio forwards calls to FS to NAT FS sends back INVITE to Kamailio Kamailio will dial-out user. Bob calls Alice Bob ==INVITE ==> Kamailio ==INVITE==> FS ==INVITE==> Kamailio ==INVITE ==> Alice How can I achieve this scenario? Can you please direct me to any documentation which is available? Thanks, Ramu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/44a01947/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Mar 6 09:05:44 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Mar 2009 17:05:44 -0000 Subject: [Freeswitch-users] Setting External IP In-Reply-To: <191c3a030903060623p16dabe4bw5f4ed1fb2d196ca4@mail.gmail.com> References: <191c3a030903060623p16dabe4bw5f4ed1fb2d196ca4@mail.gmail.com> Message-ID: Well Here's my problem From: "FreeSWITCH" ;tag=yge6eNm7a7B0r To: CSeq: 112019702 INVITE Contact: I need to change the external IP value in the contact field to a specific IP for this gateway as I'm losing the BYE message. Is there some way of manipulating this value for a given GW? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 06 March 2009 14:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting External IP gateways are children of profiles so if you need them to be separate you need to make 2 profiles and run the other one on another IP or another port. On Fri, Mar 6, 2009 at 5:04 AM, Nik Middleton wrote: Hi Guys, In External.xml in sip profiles I have Can I override these for a given gateway profile? I have one gateway that's expecting a local routed IP address due to the way that it's routed, but the other one expects the public IP, hence the need to make it gateway specific Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/5ad5b996/attachment-0002.html From lukasz at czerpak.eu Fri Mar 6 01:44:23 2009 From: lukasz at czerpak.eu (=?ISO-8859-2?Q?=A3ukasz_Czerpak?=) Date: Fri, 06 Mar 2009 10:44:23 +0100 Subject: [Freeswitch-users] Contact header on INVITE Message-ID: <49B0F077.6080908@czerpak.eu> Hi, The INVITE request from my FreeSWITCH contains (1.2.3.4 - my public ip): Contact: but voip provider expects: Contact: Is there any posiibility to change the Contact header value in gateway/dialplan configuration? Thanks, -- ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] From palletboy at gmail.com Fri Mar 6 09:08:30 2009 From: palletboy at gmail.com (J. G.) Date: Fri, 6 Mar 2009 12:08:30 -0500 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> Message-ID: <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> http://lmgtfy.com/?q=FreeSwitch+as+a+PBX On Fri, Mar 6, 2009 at 1:55 AM, Vikas Sharma wrote: > HI all, > I am new to this. > Can anybody tell me that freeSwitch can be used as PBX indecently? > Can it be integrated with other pbx as a media server? > If yes, what features it has as a media server? > > Thnax for any help. > > -- > vikas sharma > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/b830f297/attachment-0002.html From red.rain.seven at gmail.com Fri Mar 6 09:10:36 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Fri, 6 Mar 2009 09:10:36 -0800 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Message-ID: <59ad9ca10903060910u179b854evfe26ccfdfc70ba90@mail.gmail.com> Check out Shuttle XPC, they have a room for video card and a PCI slot. But you have to think a about reliability when deployed in business environment. I am using this as my home server. On Fri, Mar 6, 2009 at 8:30 AM, Fred wrote: > Hello > > I'm looking for a small, lunchbox-like PC to build a small-form > factor CRM server to sell to small companies. Ideally, it should have > one PCI slot so that I can stick a voice card to connect to an analog > phone line and run FreeSwitch as well. > > I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it > doesn't have room for a PCI slot, and I'm concerned about its performance. > > I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit > pricey, and might also not be fast enough to act as a server. > > Are there brands/models you think I should look at? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/253eb70a/attachment-0002.html From edpimentl at gmail.com Fri Mar 6 09:18:03 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 6 Mar 2009 12:18:03 -0500 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Message-ID: <9dc4a1670903060918r3aec6d53pa02f65f5140a65a9@mail.gmail.com> http://www.logicsupply.com/system_solutions http://www.advantech.com/eplatform/ -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/bcb3e442/attachment-0002.html From kristian.kielhofner at gmail.com Fri Mar 6 09:17:31 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Mar 2009 12:17:31 -0500 Subject: [Freeswitch-users] Setting External IP Message-ID: <8nRt8fFYnJQv.K5fw5rGF@smtp.gmail.com> I've never tried it but Linux routing should be able to help you with this. Create a new profile with the different address. Make sure the aliases on your network interface have a /32 netmask. Create a static route to that peer using the right outbound interface/address. Anything more than this will require iptables/iproute ip policy routing foo. Hopefully you can avoid all of that... -- Kristian Kielhofner http://blog.krisk.org -original message- Subject: [Freeswitch-users] Setting External IP From: "Nik Middleton" Date: 03/06/2009 6:05 AM Hi Guys, In External.xml in sip profiles I have Can I override these for a given gateway profile? I have one gateway that's expecting a local routed IP address due to the way that it's routed, but the other one expects the public IP, hence the need to make it gateway specific Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pablosaro at gmail.com Fri Mar 6 09:20:44 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 15:20:44 -0200 Subject: [Freeswitch-users] FS crashed Message-ID: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> Hi there, A few minutes ago, one of my FS servers crashed during normal usage (no heavy load). The hardware is Dell 1950, if you need further details please let me know. What I found on my /var/log/messages is the following: Mar 6 14:30:49 konference01 kernel: freeswitch[23798] general protection ip:7fae144e222f sp:40750540 error:0 in libc-2.7.so[7fae1446d000+14c000] I'm not familiarized with this kind of errors, but as far as I know a "general protection" occurs when a process tries to access a memory address not owned by it (probably I'm saying bullshit). The FS log does not report anything abnormal. I'm running FS svn rev 11279. Does anyone know what could happened? Thanks Pablo From nik.middleton at noblesolutions.co.uk Fri Mar 6 09:20:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Mar 2009 17:20:53 -0000 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <59ad9ca10903060910u179b854evfe26ccfdfc70ba90@mail.gmail.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> <59ad9ca10903060910u179b854evfe26ccfdfc70ba90@mail.gmail.com> Message-ID: We use the VIA mini ITX boards. Great for small offices and very stable with various fan-less options Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Henry Huang Sent: 06 March 2009 17:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Lunchbox-type PC as small server? Check out Shuttle XPC, they have a room for video card and a PCI slot. But you have to think a about reliability when deployed in business environment. I am using this as my home server. On Fri, Mar 6, 2009 at 8:30 AM, Fred wrote: Hello I'm looking for a small, lunchbox-like PC to build a small-form factor CRM server to sell to small companies. Ideally, it should have one PCI slot so that I can stick a voice card to connect to an analog phone line and run FreeSwitch as well. I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it doesn't have room for a PCI slot, and I'm concerned about its performance. I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit pricey, and might also not be fast enough to act as a server. Are there brands/models you think I should look at? Thank you. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/0099e6a6/attachment-0002.html From gmaruzz at celliax.org Fri Mar 6 09:31:44 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 6 Mar 2009 18:31:44 +0100 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> Message-ID: <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> On Fri, Mar 6, 2009 at 6:08 PM, J. G. wrote: > http://lmgtfy.com/?q=FreeSwitch+as+a+PBX wow JG, that's pretty cool! From pablosaro at gmail.com Fri Mar 6 10:15:00 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 16:15:00 -0200 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> Message-ID: <247f8100903061015x662a1db1q78e9c0fd5a482025@mail.gmail.com> LOL On Fri, Mar 6, 2009 at 3:31 PM, Giovanni Maruzzelli wrote: > On Fri, Mar 6, 2009 at 6:08 PM, J. G. wrote: >> http://lmgtfy.com/?q=FreeSwitch+as+a+PBX > > wow JG, that's pretty cool! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Mar 6 10:19:25 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Mar 2009 12:19:25 -0600 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <247f8100903061015x662a1db1q78e9c0fd5a482025@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> <247f8100903061015x662a1db1q78e9c0fd5a482025@mail.gmail.com> Message-ID: The current default config comes out of the box as a 20 extension PBX config with various features including voicemail and conferencing. /b On Mar 6, 2009, at 12:15 PM, Pablo Hernan Saro wrote: > LOL > > On Fri, Mar 6, 2009 at 3:31 PM, Giovanni Maruzzelli > wrote: >> On Fri, Mar 6, 2009 at 6:08 PM, J. G. wrote: >>> http://lmgtfy.com/?q=FreeSwitch+as+a+PBX >> >> wow JG, that's pretty cool! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/9501950b/attachment-0002.html From msc at freeswitch.org Fri Mar 6 10:31:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Mar 2009 10:31:23 -0800 Subject: [Freeswitch-users] FS crashed In-Reply-To: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> References: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> Message-ID: <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> > The FS log does not report anything abnormal. I'm running FS svn rev 11279. > Does anyone know what could happened? Well, you're about 1200 revs behind current SVN. A lot has been improved in the past month or two, so definitely get yourself to the latest SVN. -MC From msc at freeswitch.org Fri Mar 6 10:42:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Mar 2009 10:42:34 -0800 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> <3093591d0903060908h611cf789x45708810795b45b7@mail.gmail.com> <7b197bef0903060931y58c4629fy8acf0795de074056@mail.gmail.com> <247f8100903061015x662a1db1q78e9c0fd5a482025@mail.gmail.com> Message-ID: <87f2f3b90903061042v771c271h5c8789c2c78ff3d9@mail.gmail.com> On Fri, Mar 6, 2009 at 10:19 AM, Brian West wrote: > The current default config comes out of the box as a 20 extension PBX config > with various features including voicemail and conferencing. > /b And, unlike many other systems, you don't have to pay to increase the number of extensions on the system. In conf/directory/default/ there are 20 xml files: 1000.xml ... 1019.xml You can create new extensions by creating new xml files here. Then locate the "Local_Extension" in conf/dialplan/default.xml and modify the regex, which by default is this: If you want 1000~1299 to be local extensions then do this: too easy... -MC From woof at nortel.com Fri Mar 6 11:09:13 2009 From: woof at nortel.com (Andy Spitzer) Date: Fri, 06 Mar 2009 14:09:13 -0500 Subject: [Freeswitch-users] About FreeSwitch In-Reply-To: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> References: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com> Message-ID: Woof! On Fri, 06 Mar 2009 01:55:43 -0500, Vikas Sharma wrote: > Can it be integrated with other pbx as a media server? Yep, it sure can: http://sipx-wiki.calivia.com/index.php/SipXivr --Woof! From pablosaro at gmail.com Fri Mar 6 11:37:03 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 17:37:03 -0200 Subject: [Freeswitch-users] FS crashed In-Reply-To: <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> References: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> Message-ID: <247f8100903061137m405fa3act93aa2875a8136df4@mail.gmail.com> Thanks for your answer Michael. What do you recommend for production environments: the latest SVN or 1.0.3 tarball? Pablo On Fri, Mar 6, 2009 at 4:31 PM, Michael Collins wrote: >> The FS log does not report anything abnormal. I'm running FS svn rev 11279. >> Does anyone know what could happened? > > Well, you're about 1200 revs behind current SVN. A lot has been > improved in the past month or two, so definitely get yourself to the > latest SVN. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Mar 6 11:40:49 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Mar 2009 13:40:49 -0600 Subject: [Freeswitch-users] FS crashed In-Reply-To: <247f8100903061137m405fa3act93aa2875a8136df4@mail.gmail.com> References: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> <247f8100903061137m405fa3act93aa2875a8136df4@mail.gmail.com> Message-ID: latest SVN is what I would give a try. /b On Mar 6, 2009, at 1:37 PM, Pablo Hernan Saro wrote: > Thanks for your answer Michael. > What do you recommend for production environments: the latest SVN or > 1.0.3 tarball? > > Pablo From pablosaro at gmail.com Fri Mar 6 11:44:05 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 6 Mar 2009 17:44:05 -0200 Subject: [Freeswitch-users] FS crashed In-Reply-To: References: <247f8100903060920s5a5b1e95i659f2da126b5aa4a@mail.gmail.com> <87f2f3b90903061031m6a20ae99u22d3d5de180ed844@mail.gmail.com> <247f8100903061137m405fa3act93aa2875a8136df4@mail.gmail.com> Message-ID: <247f8100903061144q804d848i8476a5168860b8a2@mail.gmail.com> thx Brian. On Fri, Mar 6, 2009 at 5:40 PM, Brian West wrote: > latest ?SVN is what I would give a try. > > /b > > On Mar 6, 2009, at 1:37 PM, Pablo Hernan Saro wrote: > >> Thanks for your answer Michael. >> What do you recommend for production environments: the latest SVN or >> 1.0.3 tarball? >> >> Pablo > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From hads at nice.net.nz Fri Mar 6 11:57:34 2009 From: hads at nice.net.nz (Hadley Rich) Date: Sat, 7 Mar 2009 08:57:34 +1300 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Message-ID: <200903070857.34614.hads@nice.net.nz> On Saturday 07 March 2009 05:30:30 Fred wrote: > I'm looking for a small, lunchbox-like PC to build a small-form > factor CRM server to sell to small companies. Ideally, it should have > one PCI slot so that I can stick a voice card to connect to an analog > phone line and run FreeSwitch as well. > > I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it > doesn't have room for a PCI slot, and I'm concerned about its performance. > > I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit > pricey, and might also not be fast enough to act as a server. > > Are there brands/models you think I should look at? The Intel range of ATOM Mini-ITX boards. http://www.intel.com/products/desktop/motherboards/D945GCLF/D945GCLF- overview.htm hads -- http://nicegear.co.nz New Zealands Open Source Hardware Supplier From kristian.kielhofner at gmail.com Fri Mar 6 14:01:56 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Mar 2009 17:01:56 -0500 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> References: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> Message-ID: <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> Why not just provide a kickstart file on freeswitch.org? It's pretty easy to pass them to the installer over the network and/or add them onto existing ISOs and other bootable media... I'd be happy to write/maintain such a thing. After all, I use them myself! ;) On Fri, Mar 6, 2009 at 9:00 AM, Anthony Minessale wrote: > We are considering asking CentOS to make a "FS cut" set of packages ideal > for a telephony server with one install choice. > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From stevecrozz at gmail.com Fri Mar 6 14:07:55 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Fri, 6 Mar 2009 14:07:55 -0800 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> References: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> Message-ID: <11990ade0903061407t1b262df5uce7aa1bf53015143@mail.gmail.com> I wasn't going to say anything, but since somebody already mentioned ubuntu, I'll add that I'm using Hardy Heron LTS as well. I haven't had a single issue with that, I think it was a great choice. --Stephen On Fri, Mar 6, 2009 at 2:01 PM, Kristian Kielhofner wrote: > Why not just provide a kickstart file on freeswitch.org? ?It's pretty > easy to pass them to the installer over the network and/or add them > onto existing ISOs and other bootable media... > > I'd be happy to write/maintain such a thing. ?After all, I use them myself! ;) > > On Fri, Mar 6, 2009 at 9:00 AM, Anthony Minessale > wrote: >> We are considering asking CentOS to make a "FS cut" set of packages ideal >> for a telephony server with one install choice. >> > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Mar 6 14:28:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Mar 2009 14:28:23 -0800 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> References: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> Message-ID: <87f2f3b90903061428k30028bb9ldca1ccb3349907e0@mail.gmail.com> On Fri, Mar 6, 2009 at 2:01 PM, Kristian Kielhofner wrote: > Why not just provide a kickstart file on freeswitch.org? ?It's pretty > easy to pass them to the installer over the network and/or add them > onto existing ISOs and other bootable media... > > I'd be happy to write/maintain such a thing. ?After all, I use them myself! ;) You're hired! :) -MC From gcd at i.ph Fri Mar 6 15:03:41 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sat, 7 Mar 2009 07:03:41 +0800 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> Message-ID: <7d0bfd8c0903061503s27df5706q6671fbec46bd3dc6@mail.gmail.com> take a look at intel atom mobo d945gclf2 (dual-core). it has one PCI slot and an S-Video and VGA video ports. rhino is using this platform. On Sat, Mar 7, 2009 at 12:30 AM, Fred wrote: > Hello > > I'm looking for a small, lunchbox-like PC to build a small-form > factor CRM server to sell to small companies. Ideally, it should have > one PCI slot so that I can stick a voice card to connect to an analog > phone line and run FreeSwitch as well. > > I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it > doesn't have room for a PCI slot, and I'm concerned about its performance. > > I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit > pricey, and might also not be fast enough to act as a server. > > Are there brands/models you think I should look at? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/1b7cc5e6/attachment-0002.html From edpimentl at gmail.com Fri Mar 6 16:20:42 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 6 Mar 2009 19:20:42 -0500 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <7d0bfd8c0903061503s27df5706q6671fbec46bd3dc6@mail.gmail.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> <7d0bfd8c0903061503s27df5706q6671fbec46bd3dc6@mail.gmail.com> Message-ID: <9dc4a1670903061620n443fe2c5g6e26296ce1cb33a0@mail.gmail.com> If using Intel D945..... MoBo.. then check this out http://www.mini-box.com/Intel-D945GCLF2-Mini-ITX-Motherboard http://www.mini-box.com/M350-universal-mini-itx-enclosure;jsessionid=0a0101421f43385f65a16e0c426987a8e1ec3d627373.e3eSc3iSaN0Le34Pa38Ta38Pahf0 -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/eceac58c/attachment-0002.html From casteven at gmail.com Fri Mar 6 17:16:36 2009 From: casteven at gmail.com (Campbell Steven) Date: Sat, 07 Mar 2009 14:16:36 +1300 Subject: [Freeswitch-users] Lunchbox-type PC as small server? In-Reply-To: <9dc4a1670903061620n443fe2c5g6e26296ce1cb33a0@mail.gmail.com> References: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com> <7d0bfd8c0903061503s27df5706q6671fbec46bd3dc6@mail.gmail.com> <9dc4a1670903061620n443fe2c5g6e26296ce1cb33a0@mail.gmail.com> Message-ID: <1236388596.19995.6819.camel@macmini> I second the advantech boxes someone mentioned further up the thread, we have a few of the ARK-3000 series out there and they work great. http://www.advantech.com/products/Four-LAN-Ports-Compact-System/mod_1-2JKEGT.aspx Campbell On Fri, 2009-03-06 at 19:20 -0500, EdPimentl wrote: > If using Intel D945..... MoBo.. then check this out > http://www.mini-box.com/Intel-D945GCLF2-Mini-ITX-Motherboard > http://www.mini-box.com/M350-universal-mini-itx-enclosure;jsessionid=0a0101421f43385f65a16e0c426987a8e1ec3d627373.e3eSc3iSaN0Le34Pa38Ta38Pahf0 > > -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/9814faf5/attachment-0002.html From jaybinks at gmail.com Fri Mar 6 18:41:52 2009 From: jaybinks at gmail.com (jay binks) Date: Sat, 7 Mar 2009 12:41:52 +1000 Subject: [Freeswitch-users] Prefered Linux Distro to run FS on In-Reply-To: <11990ade0903061407t1b262df5uce7aa1bf53015143@mail.gmail.com> References: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com> <2d9149cd0903061401m84faa4as60cdc71e41214c1e@mail.gmail.com> <11990ade0903061407t1b262df5uce7aa1bf53015143@mail.gmail.com> Message-ID: I personally like and use Debian .. all my boxes are debian 4... havnt looked at using debian 5 yet. Jay On Sat, Mar 7, 2009 at 8:07 AM, Stephen Crosby wrote: > I wasn't going to say anything, but since somebody already mentioned > ubuntu, I'll add that I'm using Hardy Heron LTS as well. I haven't had > a single issue with that, I think it was a great choice. > > --Stephen > > On Fri, Mar 6, 2009 at 2:01 PM, Kristian Kielhofner > wrote: > > Why not just provide a kickstart file on freeswitch.org? It's pretty > > easy to pass them to the installer over the network and/or add them > > onto existing ISOs and other bootable media... > > > > I'd be happy to write/maintain such a thing. After all, I use them > myself! ;) > > > > On Fri, Mar 6, 2009 at 9:00 AM, Anthony Minessale > > wrote: > >> We are considering asking CentOS to make a "FS cut" set of packages > ideal > >> for a telephony server with one install choice. > >> > > > > > > -- > > Kristian Kielhofner > > http://blog.krisk.org > > http://www.submityoursip.com > > http://www.astlinux.org > > http://www.star2star.com > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/fbc6d0b7/attachment-0002.html From kjv at ken-ton.com Sat Mar 7 06:10:25 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Sat, 7 Mar 2009 09:10:25 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <49AFBFA6.45B7.0079.0@averyschools.net> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> Message-ID: <114ACB3A-3637-4FEA-A539-165A34DEAD09@ken-ton.com> Ben; > Can you imagine me injecting a question about a SIP profile into > that conversation?? Don't be bashful, just jump right in: > If I'm going into #freeswitch at 11pm at night, it's probably > because I really need some help with some problem I've run into > after hours. Can you imagine me injecting a question about a SIP > profile into that conversation?? > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though [23:12] Excuse me, I hate to interrupt, but I have a FreeSwitch question regarding (foo), can anyone help me with that? Point being that you're certainly not going to get any assistance if you neglect to ask for it. I'm not sure I speak for all of us, but we volunteer our time in the IRC channel, and on the Wiki. Since we're not paid to give our advice and/or professional opinions, we're going to "goof off" when there's no one there asking for help. We're also going to goof off when there's people in there that need help, but are too bashful or afraid to ask for it. It's human nature, and it will happen. It even happens in the work place. Should you doubt me, just watch around the water cooler at work. Guaranteed you'll see a couple folks engaged in conversation that is non-work-related. You'd probably have gotten your answer a lot sooner by asking for it in IRC as outlined above, rather that too whine about the conversations of the people that volunteer their time to help you. Summary: If you need help, ask. If you don't ask, then the only outcome of that lack of action is what? I could say more, a lot more, but I think I've made my point. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-448-3009 x0 On Mar 5, 2009, at 12:03 PM, Ben Holtsclaw wrote: > I agree with Harry. I do not like the mailing list. Those that do > like the mailing list always advocate Nabble. For those that > advocate that solution, do you even realize that you can't post on > Nabble unless you are subscribed to the mailing list? I am also not > a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in > years! I think I uninstalled my IRC client at the same time I > uninstalled my NNTP reader. Most of the time, I actually find it > difficult to obtain support in the #freeswitch channel. Once you ask > the question, if somebody doesn't happen to be there that knows the > answer, then you're screwed. How many times have I asked a question > only to wait 30 seconds and then see, "anthm has joined > #freeswitch." Crap...do I ask the question again? I have found the > conversation in #openzap to be much more focused. Thank goodness I'm > using that module! In that channel, I never see conversations about > cd burners, somebody's girlfriend in South America, or off color > jokes about someone's sexual proclivity. And because I know I'll get > flamed for saying that, just look at this: > > [23:10] <{tasker}> me, too, but i'm a different animal > [23:10] <{tasker}> in NY and in Miami i went nutz > [23:10] lol > [23:10] * jefferai is now known as lollerai > [23:10] yeah i love her > [23:10] <{tasker}> latinas everywhere > [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left > #freeswitch > [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection > timed out)) > [23:11] here its blond blue eyed girls > [23:11] * lollerai is now known as lolferai > [23:11] brazilians... hopefully she's hot. i've seen some > pretty dodgy looking chicks from there > [23:11] <{tasker}> diego: go back, buy her a ticket and tell her, > "yeaaaaaaaaah, it's just for a few days" > [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch > [23:11] <{tasker}> blonde / blue eyes are overrated > [23:11] <{tasker}> give me a latina any day > [23:11] best thing around here though > > If I'm going into #freeswitch at 11pm at night, it's probably > because I really need some help with some problem I've run into > after hours. Can you imagine me injecting a question about a SIP > profile into that conversation?? > > ALL that aside... I'm willing to use a carrier pigeon if that's the > way the three primary developers wish to communicate. They have been > instrumental in getting my project where it is today. You know the > saying... beggars can't be choosers. > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > >>> On 3/5/2009 at 5:14 AM, "J Mann/Harry" > wrote: > No, I've yet to contribute anything, I barely have my system doing > what I want. But I REALLY love Freeswitch and I want to see it BURY > Asterisk. (Windows server user here) > > I've been struggling with the XML configs, trying to figure out what > does what and where! That's fine, I'm used to it. What I'm NOT used to > is the total lack of a forum-based community to join and participate > in! Where can users SHARE their configs, help each other, learn from > each others mistakes? No DEV forum? I'm speechless. > > Let's be serious guys, mailing lists went out in the 90s! IRC is "meh" > at best! I want stickies, a forum for noobs, converts, a dev forum... > So on... > > "WELCOME To FreeSWITCH!" > > Am I asking too much here? A FORUM? > > I can't see how you can spark interest when we're so sorely lacking > the most basic and widely used community environments on the net! > > HELP SOMEBODY? Install SMF ASAP! > http://www.simplemachines.org > > BTW: I hate people who voice problems without offering viable > solutions in the process... Disgusting! So if someone can offer up a > simple hosting account, Control Panel 10, Windows Linux whatever... > I'll be more than happy to have SMF setup and receiving user > registrations within 24 hours! I've done it dozens of times before. I > will then gladly turn over the keys to the kingdom to the "powers that > be" and take a backseat, being simply a happy user from that point on! > > Please folks, please. I'm dying over here and I'm sure I'm not alone! > I'm searching Google and finding nothing!! FORUMS! > > Harry (email me here) > switchserver at gmail.com (my FS email) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/57b95830/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/57b95830/attachment-0002.bin From dave at 3c.co.uk Fri Mar 6 06:25:59 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 06 Mar 2009 07:25:59 -0700 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B0D603.502@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> Message-ID: <49B13277.9090505@3c.co.uk> Hi Rod, You can set it directly: ;screen=yes;privacy=off]]> --Dave > using these functions like this did nothing on the SIP INVITE packet :'( > > seven wrote: > >> try >> bridge >> ({effective_caller_id_name >> ="your_name",effective_caller_id_number="0000"}sofia/b-leg) >> >> On Mar 5, 2009, at 9:00 PM, rod wrote: >> >> >> >>> the A leg invite looks like this: >>> From: "Anonymous" >>> >>> it has been rewritten like this: >>> From: "Anonymous" >>> >>> rod >>> >>> rod wrote: >>> >>> >>>> Hi Brian, >>>> >>>> if I use the function effective_caller_id_number with my INVITE, I >>>> get this: >>>> >>>> From: "Anonymous" ;tag=17geyFjX5p0gS. >>>> >>>> this is not exactly what I'm looking for :p >>>> >>>> rod >>>> >>>> >>>> Brian West wrote: >>>> >>>> >>>> >>>>> Well this depends on how you're placing the call.. if its a standard >>>>> bridge you can on the A-Leg set >>>>> "effective_caller_id_number=000${caller_id_number}" before you call >>>>> bridge. >>>>> >>>>> Is the from already in the correct format? >>>>> >>>>> /b >>>>> >>>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> Dear list, >>>>>> >>>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>>> only in >>>>>> this header. >>>>>> >>>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>>> this >>>>>> example) in the RPID header : >>>>>> >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 123456 >>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> should become: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 000123456 >>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> But the From field should remain unchanged. >>>>>> >>>>>> And how to strip this prefix: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 000123456 >>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> should become: >>>>>> From: Anonymous;tag=1208367 >>>>>> Remote-Party-ID: >>>>>> >>>>> 123456 >>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>> >>>>>> >>>>>> regards. >>>>>> >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090306/036723ce/attachment-0002.html From anthony.minessale at gmail.com Sat Mar 7 08:05:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 7 Mar 2009 10:05:37 -0600 Subject: [Freeswitch-users] Contact header on INVITE In-Reply-To: <49B0F077.6080908@czerpak.eu> References: <49B0F077.6080908@czerpak.eu> Message-ID: <191c3a030903070805y1a8bdcf6j4be5f0c165f907af@mail.gmail.com> I added the param extension-in-contact (true/false) to latest trunk. Be aware it's broken behavior for your provider to demand a certain username in the contact. The reason we put gw+ in the contact user is it's the only guaranteed way to find out what gateway the call was associated with. We can of course put it in the params on the contact but we know many sip endpoints eat off all the params from your contact when you register. So we use the contact user to avoid this issue. If you use this new parameter and set it to true, the contact will be whatever is in the "extension" parameter which defaults to the same as the username, so just setting this parameter should do what you want but it will also disable our ability to associate the call with the gateway it was placed from. 2009/3/6 ?ukasz Czerpak > Hi, > > The INVITE request from my FreeSWITCH contains (1.2.3.4 - my public ip): > > Contact: > > but voip provider expects: > > Contact: > > > Is there any posiibility to change the Contact header value in > gateway/dialplan configuration? > > > Thanks, > > -- > ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/875723ea/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sat Mar 7 15:37:23 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 7 Mar 2009 23:37:23 -0000 Subject: [Freeswitch-users] Getting a sip trace on the console Message-ID: Hi Guys, I'm trying to debug some SIP messaging issues. Is there a way of doing the Asterisk equivalent of SIP Debug so I can see what's being sent? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/cc90119a/attachment-0002.html From jason at jasonjgw.net Sat Mar 7 15:56:49 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 8 Mar 2009 10:56:49 +1100 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: Message-ID: <20090307235649.GA10047@jdc.jasonjgw.net> Nik Middleton wrote: > I'm trying to debug some SIP messaging issues. Is there a way of doing > the Asterisk equivalent of SIP Debug so I can see what's being sent? http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP You can also use the sofia loglevel command from fs_cli, e.g., sofia loglevel 9 which will give you everything but a SIP trace. For the SIP trace, you'll need to set the environment variables as documented on the above page, or turn on tracing in the SIP profile configuration. From hads at nice.net.nz Sat Mar 7 16:00:15 2009 From: hads at nice.net.nz (Hadley Rich) Date: Sun, 8 Mar 2009 13:00:15 +1300 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: Message-ID: <200903081300.15833.hads@nice.net.nz> On Sunday 08 March 2009 12:37:23 Nik Middleton wrote: > I'm trying to debug some SIP messaging issues. Is there a way of doing > the Asterisk equivalent of SIP Debug so I can see what's being sent? TPORT_LOG=1 ./freeswitch will do it for you. http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP hads -- http://nicegear.co.nz New Zealands Open Source Hardware Supplier From rdmitry0911 at yandex.ru Sat Mar 7 12:12:28 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Sat, 7 Mar 2009 12:12:28 -0800 (PST) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <22339162.post@talk.nabble.com> References: <22339162.post@talk.nabble.com> Message-ID: <22391395.post@talk.nabble.com> Hi, I found one more interesting thing related to this problem. It seems, that skype protocol signaling depends on skype name used in skypiax. I was able to find couple of names, which work fine. However if I copy all settings from these lucky skype profiles to other profiles, which don't work, they do not become working anyway. In any event, I think "TRANSFERRED" and "TRANSFERRING" are valid keywords in skype signaling protocol and need to be included in skypiax_signaling_read() function Best Regards, Dmitry -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22391395.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sat Mar 7 16:54:56 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Mar 2009 18:54:56 -0600 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: <200903081300.15833.hads@nice.net.nz> References: <200903081300.15833.hads@nice.net.nz> Message-ID: if you have latest SVN you can do this: sofia profile XXX siptrace on /b On Mar 7, 2009, at 6:00 PM, Hadley Rich wrote: > On Sunday 08 March 2009 12:37:23 Nik Middleton wrote: >> I'm trying to debug some SIP messaging issues. Is there a way of >> doing >> the Asterisk equivalent of SIP Debug so I can see what's being sent? > > TPORT_LOG=1 ./freeswitch will do it for you. > > http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP > > hads -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/ad842ac7/attachment-0002.html From brian at freeswitch.org Sat Mar 7 16:55:53 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Mar 2009 18:55:53 -0600 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: <200903081300.15833.hads@nice.net.nz> References: <200903081300.15833.hads@nice.net.nz> Message-ID: <4070E206-0B7D-473E-97FA-4401F220968F@freeswitch.org> Also sofia loglevel [0-9] No need to restart FS to turn those on or off. ;) again Latest SVN. /b On Mar 7, 2009, at 6:00 PM, Hadley Rich wrote: > http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/04670f7f/attachment-0002.html From mrene_lists at avgs.ca Sat Mar 7 19:08:03 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 7 Mar 2009 22:08:03 -0500 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: <4070E206-0B7D-473E-97FA-4401F220968F@freeswitch.org> References: <200903081300.15833.hads@nice.net.nz> <4070E206-0B7D-473E-97FA-4401F220968F@freeswitch.org> Message-ID: <1E8D1AD8-02BF-4D53-95C4-014F47E94E6A@avgs.ca> only sofia profile [name] siptrace [on/off] is required to get the messages on the console sofia loglevel is used to get debugging logs from sofia's internals. (the sip library) Math On 7-Mar-09, at 7:55 PM, Brian West wrote: > Also > > sofia loglevel nth_server|nua|soa|sresolv|stun> [0-9] > > No need to restart FS to turn those on or off. ;) again Latest SVN. > > /b > > > On Mar 7, 2009, at 6:00 PM, Hadley Rich wrote: > >> http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/b77a55bb/attachment-0002.html From lfurrea at gmail.com Sat Mar 7 20:40:31 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Sat, 7 Mar 2009 22:40:31 -0600 Subject: [Freeswitch-users] Please end the torment In-Reply-To: <114ACB3A-3637-4FEA-A539-165A34DEAD09@ken-ton.com> References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <114ACB3A-3637-4FEA-A539-165A34DEAD09@ken-ton.com> Message-ID: Same thing happened to me in regards IRC, I had not used it for years before getting into FS, but as a total newbie I can say that IRC is really good to get things going quickly and all those No rocket science questions we have. I have had people helping realtime looking at pastebin's and stuff, which is a really good thing. When one needs to elaborate on a question maybe mailing list/forum fits better, but what I know in my experience is that eventually you will get the answer thanks to the efforts of the community with the current set of tools. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090307/fe5d075f/attachment-0002.html From jason at jasonjgw.net Sat Mar 7 22:00:36 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 8 Mar 2009 17:00:36 +1100 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: <200903081300.15833.hads@nice.net.nz> Message-ID: <20090308060036.GA19133@jdc.jasonjgw.net> Brian West wrote: > if you have latest SVN you can do this: > > sofia profile XXX siptrace on Thanks. That was enough to prompt me to recompile with rev. 12516. From diego.viola at gmail.com Sat Mar 7 21:41:20 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 8 Mar 2009 02:41:20 -0300 Subject: [Freeswitch-users] mod_nibblebill question: DB Error while updating cash! Message-ID: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> Hello, I'm trying to give mod_nibblebill a try, I compiled it and created the DB, set up ODBC, etc. I'm using MySQL. This is how I created the db: CREATE TABLE accounts ( id int NOT NULL PRIMARY KEY, name VARCHAR(255), cash double precision NOT NULL ); However when I try to make a call I get this: 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error while updating cash! I have this also on my user directory: Any ideas? Thanks. From diego.viola at gmail.com Sat Mar 7 22:28:42 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 8 Mar 2009 03:28:42 -0300 Subject: [Freeswitch-users] mod_nibblebill question: DB Error while updating cash! In-Reply-To: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> References: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> Message-ID: <86a32abc0903072228x67939d14x81a0c5e692cf67ac@mail.gmail.com> Oh, I noticed the billing actually works, it discounts from my credit but I still get that message, even if the update works. "2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error while updating cash!" Thanks. On Sun, Mar 8, 2009 at 3:41 AM, Diego Viola wrote: > Hello, > > I'm trying to give mod_nibblebill a try, I compiled it and created the > DB, set up ODBC, etc. I'm using MySQL. > > This is how I created the db: > > ?CREATE TABLE accounts > ?( > ? id int NOT NULL PRIMARY KEY, > ? name VARCHAR(255), > ? cash double precision NOT NULL > ?); > > However when I try to make a call I get this: > > 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error > while updating cash! > > I have this also on my user directory: > > ? ? ? > ? ? ? > > > Any ideas? > > Thanks. > From rdmitry0911 at yandex.ru Sun Mar 8 03:17:49 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Sun, 8 Mar 2009 03:17:49 -0700 (PDT) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name Message-ID: <22339162.post@talk.nabble.com> Hi all, I've got a strange problem with skypiax. I successfully installed freeswitch revision 12408 with skypiax and configured 2 skype channels with different names. When I try to call both names one by one or simultaneously, everything goes fine. But when I try to place a second call to the same skype name which is busy with the first call, I get the following message: 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ][skypiax1 ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized and second call can't get thru. I can hear call progress tones only. After about 5 seconds the message ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ][skypiax1 ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized occurs and I can hear only silence after that. Does anybody know what might cause such a problem? I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) Any help would be very much appreciated. Best regards, Dmitry -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Sun Mar 8 05:11:29 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Mar 2009 12:11:29 -0000 Subject: [Freeswitch-users] Freeswitch IAX support Message-ID: Hi Guys, Now that IAX has been published as an RFC (http://www.rfc-editor.org/authors/rfc5456.txt) are there any plans to support registrations? Not a moan, just curious as to the road map. A lot of my users have Asterisk PBX's using IAX and I'd love to replace my Asterisk central server with FS to better serve them. Yes I know I could get them to move to using SIP, but there's a lot of them. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090308/f00e8a82/attachment-0002.html From Prometheus001 at gmx.net Sun Mar 8 05:57:45 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 08 Mar 2009 13:57:45 +0100 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: Message-ID: <49B3C0C9.7040901@gmx.net> I use the ngrep tool on the OS console and write the output to a file: ngrep -d any port 5060 -W byline >outfile.txt Best regards Peter Nik Middleton schrieb: > > Hi Guys, > > > > I?m trying to debug some SIP messaging issues. Is there a way of > doing the Asterisk equivalent of SIP Debug so I can see what?s being sent? > > > > Regards, > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Sun Mar 8 07:00:23 2009 From: dujinfang at gmail.com (dujinfang) Date: Sun, 8 Mar 2009 22:00:23 +0800 Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <22339162.post@talk.nabble.com> References: <22339162.post@talk.nabble.com> Message-ID: I only use skypiax do outbound calls by using the ANY interface, and it works pretty well. It will be really cool if multi channels can call in to a single account. However, AFAIK, the skype client on my computer, if the second call come in the first channel will change to "hold". It does can accept other incoming calls, but only one channel can be active. I haven't play it into deep and not sure if one skype instance can transfer to another account or another instance on busy. On Mar 8, 2009, at 6:17 PM, rdmitry wrote: > > Hi all, > > I've got a strange problem with skypiax. I successfully installed > freeswitch > revision 12408 with skypiax and configured 2 skype channels with > different > names. When I try to call both names one by one or simultaneously, > everything goes fine. But when I try to place a second call to the > same > skype name which is busy with the first call, I get the following > message: > > 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 > skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ] > [skypiax1 > ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized > > and second call can't get thru. I can hear call progress tones only. > After > about 5 seconds the message > > ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 > skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 ] > [skypiax1 > ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized > > occurs and I can hear only silence after that. > > Does anybody know what might cause such a problem? > > I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) > > Any help would be very much appreciated. > > Best regards, Dmitry > > > > > > -- > View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rdmitry0911 at yandex.ru Sun Mar 8 07:20:54 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Sun, 8 Mar 2009 07:20:54 -0700 (PDT) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: References: <22339162.post@talk.nabble.com> Message-ID: <22398443.post@talk.nabble.com> Yes, it can. And as I can see it, freeswitch always gets control over second call, if it finds a free skypiax channel. The problem is that for some skype names it happens as skypiax expects it shoud happen, and for other names it happens the way skypiax doesn't expect and considers it as an exeption. For example, skype names like test-lineX, where X is a digit, work the way skypiax expects, while names like skyland-lineX don't work that way. I have no idea why. Nevertheless, even in this case, skypiax gets control over this call and can possibly manage it the way you would like. It just doesn't know what to do with this kind of skype protocol signaling. seven-7 wrote: > > ... I haven't play it into deep and not sure if one skype > instance can transfer to another account or another instance on busy. > > -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22398443.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gmaruzz at celliax.org Sun Mar 8 08:35:06 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 8 Mar 2009 16:35:06 +0100 Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <22339162.post@talk.nabble.com> References: <22339162.post@talk.nabble.com> Message-ID: <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> Ciao Dmitry, The warnings are unharmful, I've just fixed them as per svn 12524, so you will not see them anymore. But it will change nothing if there is a problem (I mean, the warnings are not the problem and are not indicating a problem). I cannot reproduce the problem, but maybe is because of the "strange name problem". It would be of great help if you do, from the FS CLI: console loglevel 9 then reproduce the problem, and then attach (attach, not copy) *all* the debug output (since beginning) to the Jira issue: http://jira.freeswitch.org/browse/MODSKYPIAX-28 Ciao for now, gm 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, Mar 8, 2009 at 11:17 AM, rdmitry wrote: > > Hi all, > > I've got a strange problem with skypiax. I successfully installed freeswitch > revision 12408 with skypiax and configured 2 skype channels with different > names. When I try to call both names one by one or simultaneously, > everything goes fine. But when I try to place a second call to the same > skype name which is busy with the first call, I get the following message: > > 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 > skypiax_signaling_read() rev 12409[(nil)|37 ? ? ][WARNINGA ?372 ?][skypiax1 > ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized > > and second call can't get thru. I can hear call progress tones only. After > about 5 seconds the message > > ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 > skypiax_signaling_read() rev 12409[(nil)|37 ? ? ][WARNINGA ?372 ?][skypiax1 > ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized > > occurs and I can hear only silence after that. > > Does anybody know what might cause such a problem? > > I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) > > Any help would be very much appreciated. > > Best regards, Dmitry > > > > > > -- > View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nik.middleton at noblesolutions.co.uk Sun Mar 8 09:03:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Mar 2009 16:03:45 -0000 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: <49B3C0C9.7040901@gmx.net> References: <49B3C0C9.7040901@gmx.net> Message-ID: That's exactly what I was looking for, many thanks Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P GMX Sent: 08 March 2009 12:58 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Getting a sip trace on the console I use the ngrep tool on the OS console and write the output to a file: ngrep -d any port 5060 -W byline >outfile.txt Best regards Peter Nik Middleton schrieb: > > Hi Guys, > > > > I'm trying to debug some SIP messaging issues. Is there a way of > doing the Asterisk equivalent of SIP Debug so I can see what's being sent? > > > > Regards, > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From kawarod at laposte.net Sun Mar 8 23:48:01 2009 From: kawarod at laposte.net (rod) Date: Mon, 09 Mar 2009 10:48:01 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B13277.9090505@3c.co.uk> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> Message-ID: <49B4BBA1.4000109@laposte.net> Hi David, already tried this :p the pbm is that this doesn' modify the RPID header, but it adds a new one so that I have 2 RPID header in the SIP INVITE :( rod David Knell wrote: > Hi Rod, > > You can set it directly: > application="set"> ;screen=yes;privacy=off]]> > > > --Dave > >> using these functions like this did nothing on the SIP INVITE packet :'( >> >> seven wrote: >> >>> try >>> bridge >>> ({effective_caller_id_name >>> ="your_name",effective_caller_id_number="0000"}sofia/b-leg) >>> >>> On Mar 5, 2009, at 9:00 PM, rod wrote: >>> >>> >>> >>>> the A leg invite looks like this: >>>> From: "Anonymous" >>>> >>>> it has been rewritten like this: >>>> From: "Anonymous" >>>> >>>> rod >>>> >>>> rod wrote: >>>> >>>> >>>>> Hi Brian, >>>>> >>>>> if I use the function effective_caller_id_number with my INVITE, I >>>>> get this: >>>>> >>>>> From: "Anonymous" ;tag=17geyFjX5p0gS. >>>>> >>>>> this is not exactly what I'm looking for :p >>>>> >>>>> rod >>>>> >>>>> >>>>> Brian West wrote: >>>>> >>>>> >>>>> >>>>>> Well this depends on how you're placing the call.. if its a standard >>>>>> bridge you can on the A-Leg set >>>>>> "effective_caller_id_number=000${caller_id_number}" before you call >>>>>> bridge. >>>>>> >>>>>> Is the from already in the correct format? >>>>>> >>>>>> /b >>>>>> >>>>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Dear list, >>>>>>> >>>>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>>>> only in >>>>>>> this header. >>>>>>> >>>>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>>>> this >>>>>>> example) in the RPID header : >>>>>>> >>>>>>> From: Anonymous;tag=1208367 >>>>>>> Remote-Party-ID: >>>>>>> >>>>>> 123456 >>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>> >>>>>>> should become: >>>>>>> From: Anonymous;tag=1208367 >>>>>>> Remote-Party-ID: >>>>>>> >>>>>> 000123456 >>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>> >>>>>>> But the From field should remain unchanged. >>>>>>> >>>>>>> And how to strip this prefix: >>>>>>> From: Anonymous;tag=1208367 >>>>>>> Remote-Party-ID: >>>>>>> >>>>>> 000123456 >>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>> >>>>>>> should become: >>>>>>> From: Anonymous;tag=1208367 >>>>>>> Remote-Party-ID: >>>>>>> >>>>>> 123456 >>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>> >>>>>>> >>>>>>> regards. >>>>>>> >>>>>>> >>>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rdmitry0911 at yandex.ru Mon Mar 9 01:32:18 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Mon, 9 Mar 2009 01:32:18 -0700 (PDT) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> References: <22339162.post@talk.nabble.com> <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> Message-ID: <22408941.post@talk.nabble.com> Hi Giovanni, I put everything you aked for in archive and attached it to the bug report at http://jira.freeswitch.org/browse/MODSKYPIAX-28 Hope it'll help to resolve this issue. Best Regards, Dmitry Giovanni Maruzzelli-3 wrote: > > Ciao Dmitry, > > The warnings are unharmful, I've just fixed them as per svn 12524, so > you will not see them anymore. But it will change nothing if there is > a problem (I mean, the warnings are not the problem and are not > indicating a problem). > > I cannot reproduce the problem, but maybe is because of the "strange > name problem". > > It would be of great help if you do, from the FS CLI: > > console loglevel 9 > > then reproduce the problem, and then attach (attach, not copy) *all* > the debug output (since beginning) to the Jira issue: > http://jira.freeswitch.org/browse/MODSKYPIAX-28 > > Ciao for now, > gm > > > > 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, Mar 8, 2009 at 11:17 AM, rdmitry wrote: >> >> Hi all, >> >> I've got a strange problem with skypiax. I successfully installed >> freeswitch >> revision 12408 with skypiax and configured 2 skype channels with >> different >> names. When I try to call both names one by one or simultaneously, >> everything goes fine. But when I try to place a second call to the same >> skype name which is busy with the first call, I get the following >> message: >> >> 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 >> skypiax_signaling_read() rev 12409[(nil)|37 ? ? ][WARNINGA ?372 >> ?][skypiax1 >> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized >> >> and second call can't get thru. I can hear call progress tones only. >> After >> about 5 seconds the message >> >> ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 >> skypiax_signaling_read() rev 12409[(nil)|37 ? ? ][WARNINGA ?372 >> ?][skypiax1 >> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized >> >> occurs and I can hear only silence after that. >> >> Does anybody know what might cause such a problem? >> >> I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) >> >> Any help would be very much appreciated. >> >> Best regards, Dmitry >> >> >> >> >> >> -- >> View this message in context: >> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22408941.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gmaruzz at celliax.org Mon Mar 9 01:37:04 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 9 Mar 2009 09:37:04 +0100 Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <22408941.post@talk.nabble.com> References: <22339162.post@talk.nabble.com> <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> <22408941.post@talk.nabble.com> Message-ID: <7b197bef0903090137k69e0acc4y3bffcfc3d364a65b@mail.gmail.com> Thank you Dmitry, I'll have a look into it this evening (6 hours from now :-) ) 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 Mon, Mar 9, 2009 at 9:32 AM, rdmitry wrote: > > Hi Giovanni, > > I put everything you aked for in archive and attached it to the bug report > at http://jira.freeswitch.org/browse/MODSKYPIAX-28 > > Hope it'll help to resolve this issue. > > Best Regards, Dmitry > > > Giovanni Maruzzelli-3 wrote: >> >> Ciao Dmitry, >> >> The warnings are unharmful, I've just fixed them as per svn 12524, so >> you will not see them anymore. But it will change nothing if there is >> a problem (I mean, the warnings are not the problem and are not >> indicating a problem). >> >> I cannot reproduce the problem, but maybe is because of the "strange >> name problem". >> >> It would be of great help if you do, from the FS CLI: >> >> console loglevel 9 >> >> then reproduce the problem, and then attach (attach, not copy) *all* >> the debug output (since beginning) to the Jira issue: >> http://jira.freeswitch.org/browse/MODSKYPIAX-28 >> >> Ciao for now, >> gm >> >> >> >> 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, Mar 8, 2009 at 11:17 AM, rdmitry wrote: >>> >>> Hi all, >>> >>> I've got a strange problem with skypiax. I successfully installed >>> freeswitch >>> revision 12408 with skypiax and configured 2 skype channels with >>> different >>> names. When I try to call both names one by one or simultaneously, >>> everything goes fine. But when I try to place a second call to the same >>> skype name which is busy with the first call, I get the following >>> message: >>> >>> 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 >>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>> ][skypiax1 >>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized >>> >>> and second call can't get thru. I can hear call progress tones only. >>> After >>> about 5 seconds the message >>> >>> ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 >>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>> ][skypiax1 >>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized >>> >>> occurs and I can hear only silence after that. >>> >>> Does anybody know what might cause such a problem? >>> >>> I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) >>> >>> Any help would be very much appreciated. >>> >>> Best regards, Dmitry >>> >>> >>> >>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22408941.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From diego.viola at gmail.com Sun Mar 8 19:30:01 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 8 Mar 2009 22:30:01 -0400 Subject: [Freeswitch-users] Getting a sip trace on the console In-Reply-To: References: <49B3C0C9.7040901@gmx.net> Message-ID: <86a32abc0903081930g47c83f91h64e074bbe015c49b@mail.gmail.com> Use SVN, or wait for the next release, fs_cli+siptrace rocks :) On Sun, Mar 8, 2009 at 12:03 PM, Nik Middleton wrote: > That's exactly what I was looking for, many thanks > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Peter P GMX > Sent: 08 March 2009 12:58 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Getting a sip trace on the console > > I use the ngrep tool on the OS console and write the output to a file: > ngrep -d any port 5060 -W byline >outfile.txt > > Best regards > Peter > > Nik Middleton schrieb: >> >> Hi Guys, >> >> >> >> I'm trying to debug some SIP messaging issues. ?Is there a way of >> doing the Asterisk equivalent of SIP Debug so I can see what's being > sent? >> >> >> >> Regards, >> >> > ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From zhaoxxqq at 163.com Sun Mar 8 23:44:02 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Mon, 9 Mar 2009 14:44:02 +0800 Subject: [Freeswitch-users] Codec problems about G7221 with Polycom IP550 Message-ID: <200903091444010311430@163.com> Hello, I'm a newbe of Freeswitch. I have tried to config Polycom's soundpoint IP550 to use wideband codecs. G722 has no problem with conference and dial 9999. but with G7221 and G7221c have problems. I have config vars.xml to add and config polycom IP 550's SIP.cfg like the attachment. Can anyone help me to confirm if my config is right. Zhao Xiaoqiang 2009-03-09 zhaoxxqq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/a936c825/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sip.cfg Type: application/octet-stream Size: 183561 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/a936c825/attachment-0002.obj From brian at freeswitch.org Mon Mar 9 06:37:10 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Mar 2009 08:37:10 -0500 Subject: [Freeswitch-users] Codec problems about G7221 with Polycom IP550 In-Reply-To: <200903091444010311430@163.com> References: <200903091444010311430@163.com> Message-ID: <3A52AFA3-1A00-4AC4-8D1B-E8F92745693A@freeswitch.org> The IP550, 650 and 670 DO NOT support any G722.1 codecs at this point... expect support for those later in the year... right now they only support G722. /b On Mar 9, 2009, at 1:44 AM, zhaoxxqq wrote: > Hello, > I'm a newbe of Freeswitch. I have tried to config Polycom's > soundpoint IP550 to use wideband codecs. G722 has no problem with > conference and dial 9999. but with G7221 and G7221c have problems. > I have config vars.xml to add data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h"/> and config > polycom IP 550's SIP.cfg like the attachment. > Can anyone help me to confirm if my config is right. > > Zhao Xiaoqiang > > 2009-03-09 > zhaoxxqq > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/dbfae3b4/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 9 08:15:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Mar 2009 10:15:06 -0500 Subject: [Freeswitch-users] Freeswitch IAX support In-Reply-To: References: Message-ID: <191c3a030903090815l48670058u7a388c31adfe7304@mail.gmail.com> Hi, This issue is that our mod_iax is using the only freely available iax2 stack. A client library that was only designed for small softphones. There is a patch in jira to add registration support but it was not done correctly and we have not had much time to work on it. We've already had to add several unappealing hacks to the code we are using now to make it threadsafe and i don't think it will scale very far and you may find it a disappointment even with registration support. I had suggested at some point that we would consider making an entire new scalable implementation of iax2 designed as a client/server library but really it's probably the place of the authors of the protocol to provide such a resource. But if they do no wish to, I estimated the cost of developing such a stack to be in the range of 25k-35k. So the short answer is we have little to no demand for it, so we have not put much effort into supporting it. On Sun, Mar 8, 2009 at 7:11 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > Now that IAX has been published as an RFC ( > http://www.rfc-editor.org/authors/rfc5456.txt) are there any plans to > support registrations? > > > > Not a moan, just curious as to the road map. > > > > A lot of my users have Asterisk PBX?s using IAX and I?d love to replace my > Asterisk central server with FS to better serve them. Yes I know I could get > them to move to using SIP, but there?s a lot of them. > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/b466b87c/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 9 08:18:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Mar 2009 10:18:01 -0500 Subject: [Freeswitch-users] mod_nibblebill question: DB Error while updating cash! In-Reply-To: <86a32abc0903072228x67939d14x81a0c5e692cf67ac@mail.gmail.com> References: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> <86a32abc0903072228x67939d14x81a0c5e692cf67ac@mail.gmail.com> Message-ID: <191c3a030903090818n1694d93by7433280dc65424d0@mail.gmail.com> that means you should report it to jira not the mailing list. On Sun, Mar 8, 2009 at 1:28 AM, Diego Viola wrote: > Oh, I noticed the billing actually works, it discounts from my credit > but I still get that message, even if the update works. > > "2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error > while updating cash!" > > Thanks. > > On Sun, Mar 8, 2009 at 3:41 AM, Diego Viola wrote: > > Hello, > > > > I'm trying to give mod_nibblebill a try, I compiled it and created the > > DB, set up ODBC, etc. I'm using MySQL. > > > > This is how I created the db: > > > > CREATE TABLE accounts > > ( > > id int NOT NULL PRIMARY KEY, > > name VARCHAR(255), > > cash double precision NOT NULL > > ); > > > > However when I try to make a call I get this: > > > > 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error > > while updating cash! > > > > I have this also on my user directory: > > > > > > > > > > > > Any ideas? > > > > Thanks. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/e989a275/attachment-0002.html From helmut.kuper at ewetel.de Mon Mar 9 08:19:19 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Mar 2009 16:19:19 +0100 Subject: [Freeswitch-users] Missing Diversion header in INVITE after 302 reply Message-ID: <49B53377.4040005@ewetel.de> Hello, following scenario: -Phone A is redirected unconditionally to phone C -Phone B calls A -Phone A replys with 302 and Dieversion header -FS detects the 302 and sends out a new INVITE to C I found that FS doesnt' include the received diversion sip header into the new INVITE sent to phone C. Is there a way to configure FS so that diversion header are included? Additionally: Is there a way to inform phone A about the diversion header, so that phone A get display a hint to user? regards Helmut From anthony.minessale at gmail.com Mon Mar 9 09:51:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Mar 2009 11:51:22 -0500 Subject: [Freeswitch-users] Please end the torment In-Reply-To: References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com> <49AFBFA6.45B7.0079.0@averyschools.net> <114ACB3A-3637-4FEA-A539-165A34DEAD09@ken-ton.com> Message-ID: <191c3a030903090951n675089fbjcc25e4fdf6cb26d4@mail.gmail.com> Thank you, I appriciate that you get some benifits from our efforts. We only recommend irc because it's an easy-to-access multi user chat where we can put all of the people who need help in the same room in real time so they can help each other and we can help them. I tried not to get annoyed about the 2 cracks in previous posts from others about how old and outdated irc was. Everyone is entitled to their own cup of tea after all. But we don't have to retire protocols just because they are old? We still use SMTP and HTTP and FTP and don't mock them for their age. Conversely, I feel kinda the same way about the "web 2.0" farce where the same tired browser and js nightmares I faced in 1997 are now swept under the rug with a singe addition of a background http-get instead of actually re-inventing the wheel if you are going to bother calling it wheel 2.0 Yes you can do some cool new stuff, but not nearly as much as what you could have done in 10 years of effort towards a better way, too late now ;) But that's only my opinion i don't try to enstill it to anyone. On Sat, Mar 7, 2009 at 11:40 PM, Luis F Urrea wrote: > Same thing happened to me in regards IRC, I had not used it for years > before getting into FS, but as a total newbie I can say that IRC is really > good to get things going quickly and all those No rocket science questions > we have. I have had people helping realtime looking at pastebin's and > stuff, which is a really good thing. > > When one needs to elaborate on a question maybe mailing list/forum fits > better, but what I know in my experience is that eventually you will get the > answer thanks to the efforts of the community with the current set of tools. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/2f02133e/attachment-0002.html From sergey.kirillov at gmail.com Mon Mar 9 11:41:02 2009 From: sergey.kirillov at gmail.com (Sergey Kirillov) Date: Mon, 09 Mar 2009 20:41:02 +0200 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card Message-ID: <49B562BE.1080902@gmail.com> Hi everybody, I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work. Can somebody help to to configure it? Problem log (Incoming call): ------------ 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() Unhandled event 2 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel OpenZAP/1:1/2360012 [7473c92a-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->2360012 in context default 2009-03-06 14:58:14 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer OpenZAP/1:1/2360012 to XML[1000 at default] 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 80503820933->1000 in context default 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/opt/freeswitch/recordings/80503820933.2009-03-06-14-58-14.wav 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:1000 at 192.168.122.1:5061;transport=udp [748a2ba2-0a4e-11de-9dc3-c56d4d411902] 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:1/2360012! 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state 2009-03-06 14:58:14 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:15 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2009-03-06 14:58:26 [NOTICE] Span:0 Q.921() I frame in invalid state ignored ----------- Here are my config files --- openzap.conf -- [span wanpipe BRI_1] name => BRI_1 trunk_type => bri b-channel => 1:1-2 d-channel => 1:3 --- openzap.conf.xml --- --- wanpipe1.conf --- [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] wp1aft1 = wanpipe1, auto, API, Comment wp1aft2 = wanpipe1, auto, API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 5 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 120 LBO = 120OH TE_SIG_MODE = CCS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_HW_DTMF = NO [wp1aft1] HDLC_STREAMING = NO ACTIVE_CH = 1-15.17-31 IDLE_FLAG = 0x7E MTU = 240 MRU = 240 DATA_MUX = NO TDMV_HWEC = NO [wp1aft2] HDLC_STREAMING = YES ACTIVE_CH = 16 MTU = 1500 MRU = 1500 DATA_MUX = NO TDMV_HWEC = NO From msc at freeswitch.org Mon Mar 9 12:17:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Mar 2009 12:17:19 -0700 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card In-Reply-To: <49B562BE.1080902@gmail.com> References: <49B562BE.1080902@gmail.com> Message-ID: <87f2f3b90903091217n1b611c6cp88978d2aa1a79ba2@mail.gmail.com> > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > wp1aft1 = wanpipe1, auto, API, Comment > wp1aft2 = wanpipe1, auto, API, Comment > > [wanpipe1] > CARD_TYPE ? ? ? ? = AFT > S514CPU ? ? ? ? ? = A > CommPort ? ? ? ? ?= PRI > AUTO_PCISLOT ? ? ?= NO > PCISLOT ? ? ? ? ? = 4 > PCIBUS ? ? ? ? ? ?= 5 > FE_MEDIA ? ? ? ? ?= E1 > FE_LCODE ? ? ? ? ?= HDB3 > FE_FRAME ? ? ? ? ?= CRC4 > FE_LINE ? ? ? ? ? = 1 > TE_CLOCK ? ? ? ? ?= NORMAL > TE_REF_CLOCK ? ? ?= 0 > TE_HIGHIMPEDANCE ?= NO > TE_RX_SLEVEL ? ? ?= 120 > LBO ? ? ? ? ? ? ? = 120OH > TE_SIG_MODE ? ? ? = CCS > FE_TXTRISTATE ? ? = NO > MTU ? ? ? ? ? ? ? = 1500 > UDPPORT ? ? ? ? ? = 9000 > TTL ? ? ? ? ? ? ? = 255 > IGNORE_FRONT_END ?= NO > TDMV_HW_DTMF ? ? ?= NO > > [wp1aft1] > HDLC_STREAMING ?= NO > ACTIVE_CH ? ? ? = 1-15.17-31 > IDLE_FLAG ? ? ? = 0x7E > MTU ? ? ? ? ? ? = 240 > MRU ? ? ? ? ? ? = 240 > DATA_MUX ? ? ? ?= NO > TDMV_HWEC ? ? ? = NO > > [wp1aft2] > HDLC_STREAMING ?= YES > ACTIVE_CH ? ? ? = 16 > MTU ? ? ? ? ? ? = 1500 > MRU ? ? ? ? ? ? = 1500 > DATA_MUX ? ? ? ?= NO > TDMV_HWEC ? ? ? = NO > I'm no BRI expert but it looks to me like your wanpipe is set up for E1/EuroISDN. Where did you get this setup information? -MC From anthony.minessale at gmail.com Mon Mar 9 12:18:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Mar 2009 14:18:46 -0500 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card In-Reply-To: <49B562BE.1080902@gmail.com> References: <49B562BE.1080902@gmail.com> Message-ID: <191c3a030903091218k320bf077i1ad0461ea58f4ec9@mail.gmail.com> it's not released yet, please wait for the announcement that it has been completed sometime in the next week or 2. On Mon, Mar 9, 2009 at 1:41 PM, Sergey Kirillov wrote: > Hi everybody, > > I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work. > > Can somebody help to to configure it? > > Problem log (Incoming call): > ------------ > 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() > Unhandled event 2 > 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() > Unhandled event 2 > 2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() > Unhandled event 2 > 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel OpenZAP/1:1/2360012 > [7473c92a-0a4e-11de-9dc3-c56d4d411902] > 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 80503820933->2360012 in context default > 2009-03-06 14:58:14 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() Transfer OpenZAP/1:1/2360012 to > XML[1000 at default] > 2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing 80503820933->1000 in context default > 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx > XML features > 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 > > record_session::/opt/freeswitch/recordings/80503820933.2009-03-06-14-58-14.wav > 2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf > XML features > 2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel > sofia/internal/sip:1000 at 192.168.122.1:5061;transport=udp > [748a2ba2-0a4e-11de-9dc3-c56d4d411902] > 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state > 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state > 2009-03-06 14:58:14 [NOTICE] switch_ivr_originate.c:1588 > switch_ivr_originate() Pre-Answer OpenZAP/1:1/2360012! > 2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state > 2009-03-06 14:58:14 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received > unhandled message 125 (0x7d) > 2009-03-06 14:58:15 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received > unhandled message 125 (0x7d) > 2009-03-06 14:58:26 [NOTICE] Span:0 Q.921() I frame in invalid state > ignored > ----------- > > > Here are my config files > > --- openzap.conf -- > [span wanpipe BRI_1] > name => BRI_1 > trunk_type => bri > b-channel => 1:1-2 > d-channel => 1:3 > > > --- openzap.conf.xml --- > > > > > > > > > > > > > > > > > > > > > > --- wanpipe1.conf --- > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > wp1aft1 = wanpipe1, auto, API, Comment > wp1aft2 = wanpipe1, auto, API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 4 > PCIBUS = 5 > FE_MEDIA = E1 > FE_LCODE = HDB3 > FE_FRAME = CRC4 > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > TE_HIGHIMPEDANCE = NO > TE_RX_SLEVEL = 120 > LBO = 120OH > TE_SIG_MODE = CCS > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_HW_DTMF = NO > > [wp1aft1] > HDLC_STREAMING = NO > ACTIVE_CH = 1-15.17-31 > IDLE_FLAG = 0x7E > MTU = 240 > MRU = 240 > DATA_MUX = NO > TDMV_HWEC = NO > > [wp1aft2] > HDLC_STREAMING = YES > ACTIVE_CH = 16 > MTU = 1500 > MRU = 1500 > DATA_MUX = NO > TDMV_HWEC = NO > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/192035e8/attachment-0002.html From rdmitry0911 at yandex.ru Mon Mar 9 12:37:06 2009 From: rdmitry0911 at yandex.ru (rdmitry) Date: Mon, 9 Mar 2009 12:37:06 -0700 (PDT) Subject: [Freeswitch-users] Problem with a second incoming call to the same skype user name In-Reply-To: <7b197bef0903090137k69e0acc4y3bffcfc3d364a65b@mail.gmail.com> References: <22339162.post@talk.nabble.com> <7b197bef0903080835nadd9f6et9e84515c92b6e322@mail.gmail.com> <22408941.post@talk.nabble.com> <7b197bef0903090137k69e0acc4y3bffcfc3d364a65b@mail.gmail.com> Message-ID: <22417574.post@talk.nabble.com> Hi Giovanni, Finally I was able to manage the problem. It was my fault. I didn't realized, that the value of the "name" parameter in this line: should strictly correspond to skype name you regester in startskype.sh script. It can not be arbitrary chosen, as I thought before. After I fixed that, everything works fine. I think you can put this point into skypiax page of freeswitch wiki. Best Regards, Dmitry Giovanni Maruzzelli-3 wrote: > > Thank you Dmitry, > > I'll have a look into it this evening (6 hours from now :-) ) > > > 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 Mon, Mar 9, 2009 at 9:32 AM, rdmitry wrote: >> >> Hi Giovanni, >> >> I put everything you aked for in archive and attached it to the bug >> report >> at http://jira.freeswitch.org/browse/MODSKYPIAX-28 >> >> Hope it'll help to resolve this issue. >> >> Best Regards, Dmitry >> >> >> Giovanni Maruzzelli-3 wrote: >>> >>> Ciao Dmitry, >>> >>> The warnings are unharmful, I've just fixed them as per svn 12524, so >>> you will not see them anymore. But it will change nothing if there is >>> a problem (I mean, the warnings are not the problem and are not >>> indicating a problem). >>> >>> I cannot reproduce the problem, but maybe is because of the "strange >>> name problem". >>> >>> It would be of great help if you do, from the FS CLI: >>> >>> console loglevel 9 >>> >>> then reproduce the problem, and then attach (attach, not copy) *all* >>> the debug output (since beginning) to the Jira issue: >>> http://jira.freeswitch.org/browse/MODSKYPIAX-28 >>> >>> Ciao for now, >>> gm >>> >>> >>> >>> 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, Mar 8, 2009 at 11:17 AM, rdmitry wrote: >>>> >>>> Hi all, >>>> >>>> I've got a strange problem with skypiax. I successfully installed >>>> freeswitch >>>> revision 12408 with skypiax and configured 2 skype channels with >>>> different >>>> names. When I try to call both names one by one or simultaneously, >>>> everything goes fine. But when I try to place a second call to the same >>>> skype name which is busy with the first call, I get the following >>>> message: >>>> >>>> 2009-03-04 23:00:56 [WARNING] skypiax_protocol.c:372 >>>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>>> ][skypiax1 >>>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRING is not recognized >>>> >>>> and second call can't get thru. I can hear call progress tones only. >>>> After >>>> about 5 seconds the message >>>> >>>> ED2009-03-04 23:01:02 [WARNING] skypiax_protocol.c:372 >>>> skypiax_signaling_read() rev 12409[(nil)|37 ][WARNINGA 372 >>>> ][skypiax1 >>>> ][-1, 1, 5] skype_call: 108, STATUS: TRANSFERRED is not recognized >>>> >>>> occurs and I can hear only silence after that. >>>> >>>> Does anybody know what might cause such a problem? >>>> >>>> I'm using skype client v. 2.0.0.72-1 on ubuntu 8.04 (2.6.24-23-server) >>>> >>>> Any help would be very much appreciated. >>>> >>>> Best regards, Dmitry >>>> >>>> >>>> >>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22339162.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22408941.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Problem-with-a-second-incoming-call-to-the-same-skype-user-name-tp22339162p22417574.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From diego.viola at gmail.com Mon Mar 9 13:44:50 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 9 Mar 2009 16:44:50 -0400 Subject: [Freeswitch-users] mod_nibblebill question: DB Error while updating cash! In-Reply-To: <191c3a030903090818n1694d93by7433280dc65424d0@mail.gmail.com> References: <86a32abc0903072141u30a57f7fp5f8e567617cf0299@mail.gmail.com> <86a32abc0903072228x67939d14x81a0c5e692cf67ac@mail.gmail.com> <191c3a030903090818n1694d93by7433280dc65424d0@mail.gmail.com> Message-ID: <86a32abc0903091344k316e4f81w2ae157a6c46d52fc@mail.gmail.com> Ok, done. Thanks. On Mon, Mar 9, 2009 at 11:18 AM, Anthony Minessale wrote: > that means you should report it to jira not the mailing list. > > > On Sun, Mar 8, 2009 at 1:28 AM, Diego Viola wrote: >> >> Oh, I noticed the billing actually works, it discounts from my credit >> but I still get that message, even if the update works. >> >> "2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error >> while updating cash!" >> >> Thanks. >> >> On Sun, Mar 8, 2009 at 3:41 AM, Diego Viola wrote: >> > Hello, >> > >> > I'm trying to give mod_nibblebill a try, I compiled it and created the >> > DB, set up ODBC, etc. I'm using MySQL. >> > >> > This is how I created the db: >> > >> > ?CREATE TABLE accounts >> > ?( >> > ? id int NOT NULL PRIMARY KEY, >> > ? name VARCHAR(255), >> > ? cash double precision NOT NULL >> > ?); >> > >> > However when I try to make a call I get this: >> > >> > 2009-03-08 00:37:02 [CRIT] mod_nibblebill.c:286 bill_event() DB Error >> > while updating cash! >> > >> > I have this also on my user directory: >> > >> > ? ? ? >> > ? ? ? >> > >> > >> > Any ideas? >> > >> > Thanks. >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From switchserver at gmail.com Mon Mar 9 15:17:41 2009 From: switchserver at gmail.com (Harry FSwitch) Date: Mon, 9 Mar 2009 18:17:41 -0400 Subject: [Freeswitch-users] RPC and web admin panel for conference? Message-ID: Hi all, I'm looking to implement an admin panel much like the one used at http://conference.freeswitch.org. Now I obviously cannot login and see the "admin" part of the panel but I'm pretty sure whats in there. I have xml_rpc running and can connect via http and issue commands. I've searched the forum here and went through the wiki, found nothing that looked like a panel. I was hoping to find a panel I can just configure and implement. Does anyone have a php (I guess, seeing as I have a php server) panel they can share with me? I'm sure I can get it working for my system. The thought of attempting one on my own at THIS point seems daunting at best. Any help would be greatly appreciated! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/e87c0c41/attachment-0002.html From willbelair at yahoo.com Mon Mar 9 18:36:02 2009 From: willbelair at yahoo.com (Will Smith) Date: Mon, 9 Mar 2009 18:36:02 -0700 (PDT) Subject: [Freeswitch-users] STUN error Message-ID: <85688.97806.qm@web53606.mail.re2.yahoo.com> Hi, I have the FS worked? perfectly? under NAT. And when I moved it to a server with public IP, things getting wrong. This is the error message that I got: 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] ? -- ? I tried whatever I can think of like; set the or but still got the error. Could you please give me some guide how to fix this. ? Thanks ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/375de187/attachment-0002.html From brian at freeswitch.org Mon Mar 9 18:52:10 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Mar 2009 20:52:10 -0500 Subject: [Freeswitch-users] STUN error In-Reply-To: <85688.97806.qm@web53606.mail.re2.yahoo.com> References: <85688.97806.qm@web53606.mail.re2.yahoo.com> Message-ID: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> Sounds like DNS failure maybe... might wanna remove the ext-sip-ip and ext-rtp-ip setting out of external.xml to take care of that. west philadelfia born and raised? /b On Mar 9, 2009, at 8:36 PM, Will Smith wrote: > Hi, > I have the FS worked perfectly under NAT. And when I moved it to a > server with public IP, things getting wrong. > This is the error message that I got: > 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 > sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: > 3478 [Remote Address Error!] > > -- From jason at jasonjgw.net Mon Mar 9 18:57:54 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 10 Mar 2009 12:57:54 +1100 Subject: [Freeswitch-users] STUN error In-Reply-To: <85688.97806.qm@web53606.mail.re2.yahoo.com> References: <85688.97806.qm@web53606.mail.re2.yahoo.com> Message-ID: <20090310015754.GA11792@jdc.jasonjgw.net> Will Smith wrote: > I tried whatever I can think of like; > set the > or > but still got the error. > Could you please give me some guide how to fix this. Change external_sip_ip and external_rtp_ip settings in vars.xml or in your external SIP profile. By default these are configured to use stun:stun.freeswitch.org. From willbelair at yahoo.com Mon Mar 9 20:09:44 2009 From: willbelair at yahoo.com (Will Smith) Date: Mon, 9 Mar 2009 20:09:44 -0700 (PDT) Subject: [Freeswitch-users] STUN error In-Reply-To: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> Message-ID: <694251.83720.qm@web53610.mail.re2.yahoo.com> Thank you Brian, it works like a champ. ? Yes,?west philadelfia born and raised? --- On Mon, 3/9/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] STUN error To: freeswitch-users at lists.freeswitch.org Date: Monday, March 9, 2009, 6:52 PM Sounds like DNS failure maybe... might wanna remove the ext-sip-ip and ext-rtp-ip setting out of external.xml to take care of that. west philadelfia born and raised? /b On Mar 9, 2009, at 8:36 PM, Will Smith wrote: > Hi, > I have the FS worked perfectly under NAT. And when I moved it to a > server with public IP, things getting wrong. > This is the error message that I got: > 2009-03-09 21:31:23 [ERR] sofia_glue.c:559 > sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org: > 3478 [Remote Address Error!] > > -- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/68bcc2d5/attachment-0002.html From brian at freeswitch.org Mon Mar 9 20:39:41 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Mar 2009 22:39:41 -0500 Subject: [Freeswitch-users] STUN error In-Reply-To: <694251.83720.qm@web53610.mail.re2.yahoo.com> References: <694251.83720.qm@web53610.mail.re2.yahoo.com> Message-ID: <6BB7B147-A162-4F5C-8D66-FD7BF95DEE9B@freeswitch.org> Small joke :P Do you get that a lot? /b On Mar 9, 2009, at 10:09 PM, Will Smith wrote: > Thank you Brian, it works like a champ. > > Yes, west philadelfia born and raised? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090309/d95a67a3/attachment-0002.html From dujinfang at gmail.com Mon Mar 9 21:05:35 2009 From: dujinfang at gmail.com (seven) Date: Tue, 10 Mar 2009 12:05:35 +0800 Subject: [Freeswitch-users] Script parsing a TPORT_DUMP sip log file to Mysql Message-ID: Hi all, I wrote a ruby script. it works for me. The script is in /scripts/ contrib/seven/sip/. All ideas and suggestions are welcome! Comment in script: Now and then we need to look at sip traces to see want happened on a failed call. There are lots of ways to monitor sip messages. However, not all of them are convinient as we want. Let's say a simple example: FreeSWITCH :> originate sofia/gateways/carrier1/5550000|sofia/gateways/ carrier2/5550000|sofia/carrier3... It's hard to tell what happend if the call fails. Because it's different sip sessions. The idea is to group them in one super session and see what happend. I do this by adding an arbitary sip header to do cross reference. And by parse the sip messages to a DB we can easily show it as html. I even can build a simple graph based on the DB data: http://skitch.com/seven1240/b8xj2/voip-master-idapted You can easily add a sip header to INVITE by(I use x_interaction): FreeSWITCH :> originate {sip_h_x_interaction=TEST0001}sofia/ gateways/..... And I can get all the messages from DB: SELECT * FROM `sip_messages` WHERE (call_id IN (SELECT distinct call_id FROM sip_messages WHERE x_interaction = 'TEST0001')) ORDER BY created_at; There are two aproches to get sip packets: 1) tcpdump/tshark 2) FreeSWITCH I use the second. Note, there is no way to actually get sip messages from FS currently, but sofia-sip has the ability to log all sip messages to a disk file by using TPORT_DUMP And you can use this script to parse them to a DB. I know it hurt performance, but we don't have tons of traffic and you know there are only 5-10 messages for each sip call. While we get about 1G bytes each day in the sip log, most of them are OPTIONS/NOTIFY etc. I filtered them before inserting to DB, but it will be better if sofia- sip can filter that :) The script will monitor the log file and parse and insert to DB in real time. It's written in the Ruby on Rails framework, however, I think it can run out of Rails with or without modification. But you still need ruby and rubygems if you want to use it. on Ubuntu/Debian # apt-get install ruby rubygems # gem install mysql file-tail yaml It's just an idea, you may like to write your own tools to parse the sip log file. Also the log file need to be rotated regularly. And I think it maybe possible to store the log file on a memory disk, whatever... :) Best -Seven. From diego.viola at gmail.com Mon Mar 9 21:09:55 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 10 Mar 2009 00:09:55 -0400 Subject: [Freeswitch-users] FS and Billing, AGI emulator Message-ID: <86a32abc0903092109m4f840129o6b104a14a38cdf6f@mail.gmail.com> Hi, I wanted to use A2Billing on FS, but I noticed it uses some AGI stuff for dialling and to check how much credit the user has, etc. I heard you could use A2B by just importing the FS CDR data into it, but that wont work, so I come to the conclusion that I have no way of doing billing on FS yet. There is ASTPP but that's not complete yet, and I heard vague comments of doing an AGI emulator on top of the event socket on FS, how hard that would be? Is there a possibility to do that? How much would it cost? Thanks, Diego From sergey.kirillov at gmail.com Tue Mar 10 00:48:01 2009 From: sergey.kirillov at gmail.com (Sergey Kirillov) Date: Tue, 10 Mar 2009 09:48:01 +0200 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card Message-ID: <49B61B31.8050307@gmail.com> > > I'm no BRI expert but it looks to me like your wanpipe is set up for > E1/EuroISDN. Where did you get this setup information? > -MC > It is autoconfigured by wancfg From sergey.kirillov at gmail.com Tue Mar 10 00:53:11 2009 From: sergey.kirillov at gmail.com (Sergey Kirillov) Date: Tue, 10 Mar 2009 09:53:11 +0200 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card Message-ID: <49B61C67.2070003@gmail.com> Is there a development version that I can check now? On Mon, Mar 9, 2009, Anthony Minessale wrote: > it's not released yet, > > please wait for the announcement that it has been completed sometime in the > next week or 2. From kawarod at laposte.net Tue Mar 10 04:00:57 2009 From: kawarod at laposte.net (rod) Date: Tue, 10 Mar 2009 15:00:57 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B4BBA1.4000109@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> Message-ID: <49B64869.6080007@laposte.net> Hi all, it seems there is no way to do this :( It could be great to be able to: - decide if RPID should be present or not in the B leg for an outbound call - make RPID header fully customizable with variables - filter RPID for inbound call I saw that kokoska rokoska created a jira bounty for 50$: Make RPID SIP header optional I'll add 150$ for this if I could manage RPID as described above. Sorry to use mailing list for this, I'm unable to add a note on jira for this bounty. regards, rod rod wrote: > Hi David, > > already tried this :p > the pbm is that this doesn' modify the RPID header, but it adds a new > one so that I have 2 RPID header in the SIP INVITE :( > > rod > > David Knell wrote: > >> Hi Rod, >> >> You can set it directly: >> > application="set">> ;screen=yes;privacy=off]]> >> >> >> --Dave >> >> >>> using these functions like this did nothing on the SIP INVITE packet :'( >>> >>> seven wrote: >>> >>> >>>> try >>>> bridge >>>> ({effective_caller_id_name >>>> ="your_name",effective_caller_id_number="0000"}sofia/b-leg) >>>> >>>> On Mar 5, 2009, at 9:00 PM, rod wrote: >>>> >>>> >>>> >>>> >>>>> the A leg invite looks like this: >>>>> From: "Anonymous" >>>>> >>>>> it has been rewritten like this: >>>>> From: "Anonymous" >>>>> >>>>> rod >>>>> >>>>> rod wrote: >>>>> >>>>> >>>>> >>>>>> Hi Brian, >>>>>> >>>>>> if I use the function effective_caller_id_number with my INVITE, I >>>>>> get this: >>>>>> >>>>>> From: "Anonymous" ;tag=17geyFjX5p0gS. >>>>>> >>>>>> this is not exactly what I'm looking for :p >>>>>> >>>>>> rod >>>>>> >>>>>> >>>>>> Brian West wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Well this depends on how you're placing the call.. if its a standard >>>>>>> bridge you can on the A-Leg set >>>>>>> "effective_caller_id_number=000${caller_id_number}" before you call >>>>>>> bridge. >>>>>>> >>>>>>> Is the from already in the correct format? >>>>>>> >>>>>>> /b >>>>>>> >>>>>>> On Mar 5, 2009, at 6:12 AM, rod wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> Dear list, >>>>>>>> >>>>>>>> I'd like to rewrite the number in the Remote Party ID header and >>>>>>>> only in >>>>>>>> this header. >>>>>>>> >>>>>>>> ex: I'd like to prefix the caller ID with a prefix code (000 in >>>>>>>> this >>>>>>>> example) in the RPID header : >>>>>>>> >>>>>>>> From: Anonymous;tag=1208367 >>>>>>>> Remote-Party-ID: >>>>>>>> >>>>>>> 123456 >>>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>>> >>>>>>>> should become: >>>>>>>> From: Anonymous;tag=1208367 >>>>>>>> Remote-Party-ID: >>>>>>>> >>>>>>> 000123456 >>>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>>> >>>>>>>> But the From field should remain unchanged. >>>>>>>> >>>>>>>> And how to strip this prefix: >>>>>>>> From: Anonymous;tag=1208367 >>>>>>>> Remote-Party-ID: >>>>>>>> >>>>>>> 000123456 >>>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>>> >>>>>>>> should become: >>>>>>>> From: Anonymous;tag=1208367 >>>>>>>> Remote-Party-ID: >>>>>>>> >>>>>>> 123456 >>>>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling >>>>>>>> >>>>>>>> >>>>>>>> regards. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From kokoska.rokoska at post.cz Tue Mar 10 04:30:49 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 12:30:49 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B64869.6080007@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> Message-ID: <49B64F69.5070209@post.cz> rod napsal(a): > Hi all, > > it seems there is no way to do this :( > > It could be great to be able to: > - decide if RPID should be present or not in the B leg for an > outbound call > - make RPID header fully customizable with variables > - filter RPID for inbound call > > I saw that kokoska rokoska created a jira bounty for 50$: Make RPID SIP > header optional > > I'll add 150$ for this if I could manage RPID as described above. > Sorry to use mailing list for this, I'm unable to add a note on jira for > this bounty. > I really don't need to customize RPID (cause it is depricated header), but... Well let's add another 50 $ if it will work like you describe :-) Best regards, kokoska.rokoska From kawarod at laposte.net Tue Mar 10 05:13:18 2009 From: kawarod at laposte.net (rod) Date: Tue, 10 Mar 2009 16:13:18 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B64F69.5070209@post.cz> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> Message-ID: <49B6595E.6040700@laposte.net> Yes I know, it's deprecated but many peers still rely on this and P-Asserted-ID is not widely spread (my own experience). moreover if we could strip the RPID, we could write a new one, but It could be very convenient to get access to the fields in this header for manipulation. thanks for supporting this request :p kokoska rokoska wrote: > > > rod napsal(a): > >> Hi all, >> >> it seems there is no way to do this :( >> >> It could be great to be able to: >> - decide if RPID should be present or not in the B leg for an >> outbound call >> - make RPID header fully customizable with variables >> - filter RPID for inbound call >> >> I saw that kokoska rokoska created a jira bounty for 50$: Make RPID SIP >> header optional >> >> I'll add 150$ for this if I could manage RPID as described above. >> Sorry to use mailing list for this, I'm unable to add a note on jira for >> this bounty. >> >> > > I really don't need to customize RPID (cause it is depricated header), > but... Well let's add another 50 $ if it will work like you describe :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From Richard.Lamkin at mettoni.com Tue Mar 10 03:26:22 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Tue, 10 Mar 2009 10:26:22 -0000 Subject: [Freeswitch-users] FS stopped working when NIC connection bounced. Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804930748@nickel.mettonigroup.com> Hi All, FS stopped working when NIC connection bounced. Q1- Has anyone an explanation of what happened ? Q2 - Is there a way to configure FS not to flip to another IP on a lost network connection. Q3 - Should an IP changed event be logged at a higher level e.g. [CRITICAL] ? NIC2 [172.22.240.156] lost connection to LAN then Freeswitch flipped to NIC1[192.168.1.1]. Then NIC2 restored connection and Freeswitch flipped back to NIC2. When it flipped back no SIP connections were restored. All PBX phones lost registration and my gateways to remote switches did not recover. FS is running but not responding to CLI or SIP. There is nothing logged in the Windows event viewer at the time of the incident. Sorry but I was not running Wireshark when this happened. ========== Log extract ======== 2009-03-09 21:00:21 [INFO] mod_sofia.c:2785 general_event_handler() IP change detected [172.22.240.156]->[192.168.1.1] []->[] 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write lock external 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting for worker thread 2009-03-09 21:00:21 [NOTICE] sofia_glue.c:2923 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-03-09 21:00:21 [INFO] switch_time.c:656 switch_load_timezones() Timezone reloaded 530 definitions 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write lock internal 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting for worker thread 2009-03-09 21:11:21 [INFO] mod_sofia.c:2785 general_event_handler() IP change detected [192.168.1.1]->[172.22.240.156] []->[] 2009-03-09 21:11:21 [NOTICE] sofia_glue.c:2923 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-03-09 21:11:21 [INFO] switch_time.c:656 switch_load_timezones() Timezone reloaded 530 definitions There were no more logged events after 2009-03-09 21:11:21. The log file was checked at 2009-03-10 08:50 . All phones had lost registration permanently I tried the CLI with "sofia status" this failed to return anything and the CLI was no longer responsive. I did a CTR-C just to see what was alive [ VS2008 trapped the break and I selected continue] 2009-03-10 08:58:25 [WARNING] switch_scheduler.c:114 task_thread_loop() Task was executed late by 4 seconds 1 heartbeat (core) Still no CLI. I the killed the FS, restarted and it worked as normal ==== Background; I'm testing Freeswitch on Windows XP with Release 1.03 from the source release tar ball, running in VS2008 debug mode. The PC has two NIC's 1 - 192.168.1.1 static 2 - 172.22.240.156 DHCP I'm using Nortel switches CS1K and CS2K for my upstream gateways and Linksys SPA942 for PBX clients. I'm only using SIP <-> SIP and all sip connections are via 172.22.240.156, There are no SIP devices on 192.168.1.1. It is unlikely the event was caused by a DHCP lease renew. DHCP lease details for 172.22.240.156. Lease Obtained. . . . . . . . . . : 09 March 2009 15:27:28 Lease Expires . . . . . . . . . . : 17 March 2009 15:27:28 === Regards Richard Lamkin PS: I am a new to FS and I am very enthusiastic about it potential. ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/92ad4c3f/attachment-0002.html From kokoska.rokoska at post.cz Tue Mar 10 05:32:51 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 13:32:51 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B6595E.6040700@laposte.net> References: <49AFC1C3.9030603@laposte.net> <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org> <49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> Message-ID: <49B65DF3.9060001@post.cz> rod napsal(a): > ... if we could strip the RPID, we could write a new one, but It > could be very convenient to get access to the fields in this header for > manipulation. > Yes, rod, this is exactly why I update bounty to $100 :-) Thank you very much, rod, for support! Best regards, kokoska.rokoska From anthony.minessale at gmail.com Tue Mar 10 06:16:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 08:16:19 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B65DF3.9060001@post.cz> References: <49AFC1C3.9030603@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> Message-ID: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Latest SVN: Send no extra caller id info: {sip_cid_type=none}sofia/default/user at example.com Send RPID (default) {sip_cid_type=rpid}sofia/default/user at example.com Send P-XXX-Identity {sip_cid_type=pid}sofia/default/user at example.com Send RPID with chosen content {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ user at example.com [ Show ? ] Anthony Minessale IIadded a comment - 10/Mar/09 07:59 AM Send no extra caller id info: {sip_cid_type=none}sofia/default/ user at example.com Send RPID (default) {sip_cid_type=rpid}sofia/default/ user at example.com Send P-XXX-Identity {sip_cid_type=pid}sofia/default/ user at example.com Send RPID with chosen content {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ user at example.com Send RPID with chosen content and privacy flags (+ delimited, none to clear all flags) {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/ user at example.com [ Show ? ] Anthony Minessale IIadded a comment - 10/Mar/09 07:59 AM Send no extra caller id info: {sip_cid_type=none}sofia/default/ user at example.com Send RPID (default) {sip_cid_type=rpid}sofia/default/ user at example.com Send P-XXX-Identity {sip_cid_type=pid}sofia/default/ user at example.com Send RPID with chosen content {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ user at example.com Also the privacy app on the inbound leg controls the remaining contents of the RPID and Privacy headers. On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska wrote: > > rod napsal(a): > > ... if we could strip the RPID, we could write a new one, but It > > could be very convenient to get access to the fields in this header for > > manipulation. > > > > Yes, rod, this is exactly why I update bounty to $100 :-) > > Thank you very much, rod, for support! > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/2b6b7657/attachment-0002.html From anthony.minessale at gmail.com Tue Mar 10 06:22:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 08:22:50 -0500 Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card In-Reply-To: <49B61B31.8050307@gmail.com> References: <49B61B31.8050307@gmail.com> Message-ID: <191c3a030903100622g358ea918rb76d328f329bfbdf@mail.gmail.com> As I have already stated, it will be added to SVN as soon as it's complete. On Tue, Mar 10, 2009 at 2:48 AM, Sergey Kirillov wrote: > > > > I'm no BRI expert but it looks to me like your wanpipe is set up for > > E1/EuroISDN. Where did you get this setup information? > > -MC > > > It is autoconfigured by wancfg > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/90ebe410/attachment-0002.html From sicfslist at gmail.com Tue Mar 10 06:28:25 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 10 Mar 2009 08:28:25 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Message-ID: <35b355e90903100628m277265d7g1a9dfe1492dd2a6b@mail.gmail.com> Anthony, That is awesome. This is something that will be a BIG help. SDR > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/3008dc65/attachment-0002.html From kokoska.rokoska at post.cz Tue Mar 10 06:36:44 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 14:36:44 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Message-ID: <49B66CEC.7060002@post.cz> Anthony Minessale napsal(a): > Latest SVN: > > Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > [ Show ? ] > Anthony Minessale II > added a > comment - 10/Mar/09 07:59 AM Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > Send RPID with chosen content and privacy flags (+ delimited, none to > clear all flags) > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/user at example.com > > > > [ Show ? ] > Anthony Minessale II > added a > comment - 10/Mar/09 07:59 AM Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > Also the privacy app on the inbound leg controls the remaining contents > of the RPID and Privacy headers. > Incredible speed - like usually :-) Thank you very much, Anthony, for your work! Where should I send my bucks? Best regards, kokoska.rokoska From anthony.minessale at gmail.com Tue Mar 10 07:19:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 09:19:42 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B66CEC.7060002@post.cz> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B66CEC.7060002@post.cz> Message-ID: <191c3a030903100719w4ec5b18ei6d39871758c56231@mail.gmail.com> The paypal button on the hompege will do ;) On Tue, Mar 10, 2009 at 8:36 AM, kokoska rokoska wrote: > > > > Anthony Minessale napsal(a): > > Latest SVN: > > > > Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com user at example.com> > > > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com user at example.com> > > > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com user at example.com> > > > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > [ Show ? ] > > Anthony Minessale II > > added a > > comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > > > Send RPID with chosen content and privacy flags (+ delimited, none to > > clear all flags) > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/ > user at example.com > > > > > > > > [ Show ? ] > > Anthony Minessale II > > added a > > comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > > > > > Also the privacy app on the inbound leg controls the remaining contents > > of the RPID and Privacy headers. > > > > Incredible speed - like usually :-) > > Thank you very much, Anthony, for your work! Where should I send my bucks? > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/6b471531/attachment-0002.html From anthony.minessale at gmail.com Tue Mar 10 07:23:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 09:23:21 -0500 Subject: [Freeswitch-users] FS stopped working when NIC connection bounced. In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804930748@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804930748@nickel.mettonigroup.com> Message-ID: <191c3a030903100723u7d87cd20t8710c41ece88d2d7@mail.gmail.com> conf/autoload_configs/sofia.conf.xml uncomment out the auto-restart. On Tue, Mar 10, 2009 at 5:26 AM, Richard Lamkin wrote: > Hi All, > > > > FS stopped working when NIC connection bounced. > > > > Q1- Has anyone an explanation of what happened ? > > Q2 - Is there a way to configure FS not to flip to another IP on a lost > network connection. > > Q3 ? Should an IP changed event be logged at a higher level e.g. > [CRITICAL] ? > > > > NIC2 [172.22.240.156] lost connection to LAN then Freeswitch flipped to > NIC1[192.168.1.1]. Then NIC2 restored connection and Freeswitch flipped back > to NIC2. When it flipped back no SIP connections were restored. All PBX > phones lost registration and my gateways to remote switches did not > recover. FS is running but not responding to CLI or SIP. There is nothing > logged in the Windows event viewer at the time of the incident. Sorry but I > was not running Wireshark when this happened. > > > > ========== Log extract ======== > > > > 2009-03-09 21:00:21 [INFO] mod_sofia.c:2785 general_event_handler() IP > change detected [172.22.240.156]->[192.168.1.1] []->[] > > 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write > lock external > > 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting > for worker thread > > 2009-03-09 21:00:21 [NOTICE] sofia_glue.c:2923 > sofia_glue_restart_all_profiles() Reload XML [Success] > > 2009-03-09 21:00:21 [INFO] switch_time.c:656 switch_load_timezones() > Timezone reloaded 530 definitions > > 2009-03-09 21:00:21 [DEBUG] sofia.c:848 sofia_profile_thread_run() Write > lock internal > > 2009-03-09 21:00:21 [NOTICE] sofia.c:857 sofia_profile_thread_run() Waiting > for worker thread > > 2009-03-09 21:11:21 [INFO] mod_sofia.c:2785 general_event_handler() IP > change detected [192.168.1.1]->[172.22.240.156] []->[] > > 2009-03-09 21:11:21 [NOTICE] sofia_glue.c:2923 > sofia_glue_restart_all_profiles() Reload XML [Success] > > 2009-03-09 21:11:21 [INFO] switch_time.c:656 switch_load_timezones() > Timezone reloaded 530 definitions > > > > There were no more logged events after 2009-03-09 21:11:21. The log file > was checked at 2009-03-10 08:50 . All phones had lost registration > permanently > > I tried the CLI with ?sofia status? this failed to return anything and the > CLI was no longer responsive. > > > > I did a CTR-C just to see what was alive [ VS2008 trapped the break and I > selected continue] > > > > 2009-03-10 08:58:25 [WARNING] switch_scheduler.c:114 task_thread_loop() > Task was executed late by 4 seconds 1 heartbeat (core) > > > > Still no CLI. > > > > I the killed the FS, restarted and it worked as normal > > > > ==== > > Background; I?m testing Freeswitch on Windows XP with Release 1.03 from the > source release tar ball, running in VS2008 debug mode. > > > > The PC has two NIC?s > > 1 - 192.168.1.1 static > > 2 - 172.22.240.156 DHCP > > > > I?m using Nortel switches CS1K and CS2K for my upstream gateways and > Linksys SPA942 for PBX clients. > > > > I?m only using SIP <-> SIP and all sip connections are via 172.22.240.156, > There are no SIP devices on 192.168.1.1. > > > > It is unlikely the event was caused by a DHCP lease renew. > > > > DHCP lease details for 172.22.240.156. > > Lease Obtained. . . . . . . . . . : 09 March 2009 15:27:28 > > Lease Expires . . . . . . . . . . : 17 March 2009 15:27:28 > > > > === > > > > Regards > > > > Richard Lamkin > > > > PS: I am a new to FS and I am very enthusiastic about it potential. > > > > > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/1e30cebf/attachment-0002.html From kawarod at laposte.net Tue Mar 10 07:30:28 2009 From: kawarod at laposte.net (rod) Date: Tue, 10 Mar 2009 18:30:28 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49AFCD02.2000603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Message-ID: <49B67984.30104@laposte.net> Hi Anthony, thanks for this but I'd like to know if it's possible also to change only the caller_id_name and caller_id_number without modifying the from header. ex: with the origination variables I get this From: "test" ;tag=X4v4Kvt1B2DQF Remote-Party-ID: "test" ;party=calling;screen=yes;privacy=off and in the case of an anonymous call where from is like this on A-leg: From: Anonymous ;tag=z07m5db13cb-cbs450547977-o-5273-368294925 it is changed like this on B-leg: From: "test" ;tag=X4v4Kvt1B2DQF Moreover, in case of a bridged call, could you add the possibility to pass as is from header for an anonymous call. Running current, I have this: on A-leg: From: "Anonymous";tag=c0a80101-fed0c on B-leg: From: "Anonymous" ;tag=tpjD57U2pyrFN and I'd like B-leg to match A-leg for anonymous call as stated in RFC, anonymous at anonymous.invalid is the proposed way to handle anonymous call, I'll add 50$ more for support of this last request. thanks for your reactivity. regards, rod Anthony Minessale wrote: > Latest SVN: > > Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > > > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > > > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > > > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > [ Show ? ] > Anthony Minessale II > added > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > Send RPID with chosen content and privacy flags (+ delimited, none to > clear all flags) > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/user at example.com > > > > [ Show ? ] > Anthony Minessale II > added > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > Send RPID with chosen content > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > Also the privacy app on the inbound leg controls the remaining > contents of the RPID and Privacy headers. > > > On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska > > wrote: > > > rod napsal(a): > > ... if we could strip the RPID, we could write a new one, but It > > could be very convenient to get access to the fields in this > header for > > manipulation. > > > > Yes, rod, this is exactly why I update bounty to $100 :-) > > Thank you very much, rod, for support! > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kokoska.rokoska at post.cz Tue Mar 10 07:57:20 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 15:57:20 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100719w4ec5b18ei6d39871758c56231@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B66CEC.7060002@post.cz> <191c3a030903100719w4ec5b18ei6d39871758c56231@mail.gmail.com> Message-ID: <49B67FD0.5070105@post.cz> Done :-) kokoska.rokoska Anthony Minessale napsal(a): > The paypal button on the hompege will do ;) > From anthony.minessale at gmail.com Tue Mar 10 08:17:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 10:17:23 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B67984.30104@laposte.net> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> Message-ID: <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> ok if you are up to date you should be able to add {sip_from_uri=sip:anonymous at anonymous.invalid} to your dial string. On Tue, Mar 10, 2009 at 9:30 AM, rod wrote: > Hi Anthony, > > thanks for this but I'd like to know if it's possible also to change > only the caller_id_name and caller_id_number without modifying the from > header. > > ex: with the origination variables I get this > From: "test" > >;tag=X4v4Kvt1B2DQF > Remote-Party-ID: "test" > > >;party=calling;screen=yes;privacy=off > > and in the case of an anonymous call where from is like this on A-leg: > From: Anonymous > >;tag=z07m5db13cb-cbs450547977-o-5273-368294925 > > it is changed like this on B-leg: > From: "test" > >;tag=X4v4Kvt1B2DQF > > Moreover, in case of a bridged call, could you add the possibility to > pass as is from header for an anonymous call. > Running current, I have this: > on A-leg: > From: > "Anonymous";tag=c0a80101-fed0c > on B-leg: > From: "Anonymous" > >;tag=tpjD57U2pyrFN > > and I'd like B-leg to match A-leg for anonymous call as stated in RFC, > anonymous at anonymous.invalid is the proposed way to handle anonymous > call, I'll add 50$ more for support of this last request. > > thanks for your reactivity. > regards, > rod > > Anthony Minessale wrote: > > Latest SVN: > > > > Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > > > > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > > > > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > > > > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > [ Show ? ] > > Anthony Minessale II > > added > > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > > > Send RPID with chosen content and privacy flags (+ delimited, none to > > clear all flags) > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/ > user at example.com > > > > > > > > [ Show ? ] > > Anthony Minessale II > > added > > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/ > user at example.com > > > > > > > > Also the privacy app on the inbound leg controls the remaining > > contents of the RPID and Privacy headers. > > > > > > On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska > > > wrote: > > > > > > rod napsal(a): > > > ... if we could strip the RPID, we could write a new one, but It > > > could be very convenient to get access to the fields in this > > header for > > > manipulation. > > > > > > > Yes, rod, this is exactly why I update bounty to $100 :-) > > > > Thank you very much, rod, for support! > > > > Best regards, > > > > kokoska.rokoska > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/8bff0048/attachment-0002.html From kokoska.rokoska at post.cz Tue Mar 10 08:53:56 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Mar 2009 16:53:56 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> Message-ID: <49B68D14.5090908@post.cz> Anthony Minessale napsal(a): > ok if you are up to date you should be able to add > > {sip_from_uri=sip:anonymous at anonymous.invalid} to your dial string. > > Many thanks, Anthony, for that feature! It makes my life a lot easier :-) Best regards, kokoska.rokoska From helmut.kuper at ewetel.de Tue Mar 10 09:00:15 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 10 Mar 2009 17:00:15 +0100 Subject: [Freeswitch-users] Missing Diversion header in INVITE after 302 reply In-Reply-To: <49B53377.4040005@ewetel.de> References: <49B53377.4040005@ewetel.de> Message-ID: <49B68E8F.1010803@ewetel.de> Hello, has anybody an idea? regards Helmut On 09.03.2009 16:19, Helmut Kuper wrote: > Hello, > > following scenario: > > -Phone A is redirected unconditionally to phone C > -Phone B calls A > -Phone A replys with 302 and Dieversion header > -FS detects the 302 and sends out a new INVITE to C > > I found that FS doesn't include the received diversion sip header into > the new INVITE sent to phone C. > > Is there a way to configure FS so that diversion header are included? > > Additionally: Is there a way to inform phone A about the diversion > header, so that phone A get display a hint to user? > > regards > Helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From anthony.minessale at gmail.com Tue Mar 10 10:18:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Mar 2009 12:18:07 -0500 Subject: [Freeswitch-users] IRC pre release party Message-ID: <191c3a030903101018s3e8e0c3ai27cec27f7bb8ba23@mail.gmail.com> If you have nothing better to do drop by IRC We are up to 193 users and about to cross 200 for the first time. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/1302c16d/attachment-0002.html From msc at freeswitch.org Tue Mar 10 13:48:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Mar 2009 13:48:08 -0700 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> Message-ID: <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale wrote: > Latest SVN: > > Send no extra caller id info: > {sip_cid_type=none}sofia/default/user at example.com > > Send RPID (default) > {sip_cid_type=rpid}sofia/default/user at example.com > > Send P-XXX-Identity > {sip_cid_type=pid}sofia/default/user at example.com > > Send RPID with chosen content >{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul tuser at example.com FYI, I added this info to the channel variables page: http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type -MC From Prometheus001 at gmx.net Tue Mar 10 15:20:39 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 10 Mar 2009 23:20:39 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> Message-ID: <49B6E7B7.1090003@gmx.net> Hello, are these variables only available at call setup time or can they be changed during a call, e.g. before a call is being transferred to another destination? Best regards Peter Michael Collins schrieb: > On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale > wrote: > >> Latest SVN: >> >> Send no extra caller id info: >> {sip_cid_type=none}sofia/default/user at example.com >> >> Send RPID (default) >> {sip_cid_type=rpid}sofia/default/user at example.com >> >> Send P-XXX-Identity >> {sip_cid_type=pid}sofia/default/user at example.com >> >> Send RPID with chosen content >> {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul tuser at example.com >> > > FYI, I added this info to the channel variables page: > http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From krice at suspicious.org Tue Mar 10 15:29:30 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 10 Mar 2009 17:29:30 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B6E7B7.1090003@gmx.net> Message-ID: These should be available any time you are going to process a call thru the dialplan and call a bridge on the call > From: Peter P GMX > Reply-To: > Date: Tue, 10 Mar 2009 23:20:39 +0100 > To: > Subject: Re: [Freeswitch-users] Rewriting Remote Party ID > > Hello, > > are these variables only available at call setup time or can they be > changed during a call, e.g. before a call is being transferred to > another destination? > > Best regards > Peter > > Michael Collins schrieb: >> On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale >> wrote: >> >>> Latest SVN: >>> >>> Send no extra caller id info: >>> {sip_cid_type=none}sofia/default/user at example.com >>> >>> Send RPID (default) >>> {sip_cid_type=rpid}sofia/default/user at example.com >>> >>> Send P-XXX-Identity >>> {sip_cid_type=pid}sofia/default/user at example.com >>> >>> Send RPID with chosen content >>> {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_num >>> ber=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul >>> tuser at example.com >>> >> >> FYI, I added this info to the channel variables page: >> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type >> >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at suspicious.org Tue Mar 10 15:32:53 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 10 Mar 2009 17:32:53 -0500 Subject: [Freeswitch-users] RPC and web admin panel for conference? In-Reply-To: Message-ID: Hey, I just implemented something like this and commited it to my contrib directory (scripts/contrib/swk ) its a mixture of amf-php, ESL, and Flex... Its not complete by anymeans and you need Flex3 to compile the UI... Anyone wanting to throw some patches at it for other functionality are welcome to do so... One thing is severly lacks at this time is ANY sort of authentication...so you wouldn?t want it publically open to the world K From: Harry FSwitch Reply-To: Date: Mon, 9 Mar 2009 18:17:41 -0400 To: Subject: [Freeswitch-users] RPC and web admin panel for conference? Hi all, I'm looking to implement an admin panel much like the one used at http://conference.freeswitch.org. Now I obviously cannot login and see the "admin" part of the panel but I'm pretty sure whats in there. I have xml_rpc running and can connect via http and issue commands. I've searched the forum here and went through the wiki, found nothing that looked like a panel. I was hoping to find a panel I can just configure and implement. Does anyone have a php (I guess, seeing as I have a php server) panel they can share with me? I'm sure I can get it working for my system. The thought of attempting one on my own at THIS point seems daunting at best. Any help would be greatly appreciated! Thanks _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090310/69a3a6f0/attachment-0002.html From mszlazak at aol.com Tue Mar 10 23:46:40 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 11 Mar 2009 02:46:40 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. Message-ID: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> I have the following in my dialplan. Individually, each extension does what it's suppose to do when dialing 1111. However, if I place continue=true in the first extension then it alone gets executed and the succeeding extension does not. I thought condition=true would allow the extension afterward to execute. My test hardware for this dialplan is a single PSTN line. A call comes in that line and "myExtension" executes then hopefully hangs up to free the line. Afterward, I want "myExtension_Continued" to execute the .js application and dial out that single PSTN line. I need help in getting this scenerio to work. Thanks. ?? ? ?? ??? ? ?? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ? ?? ? ?? ??? ? ?? ??? ? ?? ??? ??? ? ?? ??? ? ?? ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/e6dc106f/attachment-0002.html From helmut.kuper at ewetel.de Wed Mar 11 01:07:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Mar 2009 09:07:02 +0100 Subject: [Freeswitch-users] Waiting for playing a file as early media Message-ID: <49B77126.2030500@ewetel.de> Hello, I'm looking for a way to play a file exactly once in early media state of a call and then do e.g. a 302 reply. When I try it via ring_ready and ringback The playback immediately stops, when I send a respond. regards Helmut From kawarod at laposte.net Wed Mar 11 01:33:33 2009 From: kawarod at laposte.net (rod) Date: Wed, 11 Mar 2009 12:33:33 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> Message-ID: <49B7775D.3030308@laposte.net> thanks a lot Anthony, I consider the work is done and I did the paypal transfer of 200$. But I'll be happy if you could do some minor changes: I used this in the dialplan: and I get this From: "test" ;tag=tQ52gmv6NyN0D. Remote-Party-ID: "test" ;party=calling;screen=yes;privacy=off. as you may see, the origination_caller_id_name has been put in the from URI, and the domain part in the remote party ID has been transformed to @anonymous.invalid. It would be perfect if I could get this instead: From: "Anonymous" ;tag=tQ52gmv6NyN0D. Remote-Party-ID: "test" ;party=calling;screen=yes;privacy=off. where 172.29.0.5 is the external IP used by FS to bridge the call in this example (but I'm sure you already know this :p) regards, rod Anthony Minessale wrote: > ok if you are up to date you should be able to add > > {sip_from_uri=sip:anonymous at anonymous.invalid} to your dial string. > > > On Tue, Mar 10, 2009 at 9:30 AM, rod > wrote: > > Hi Anthony, > > thanks for this but I'd like to know if it's possible also to change > only the caller_id_name and caller_id_number without modifying the > from > header. > > ex: with the origination variables I get this > From: "test" >;tag=X4v4Kvt1B2DQF > Remote-Party-ID: "test" > >;party=calling;screen=yes;privacy=off > > and in the case of an anonymous call where from is like this on A-leg: > From: Anonymous > ;tag=z07m5db13cb-cbs450547977-o-5273-368294925 > > it is changed like this on B-leg: > From: "test" >;tag=X4v4Kvt1B2DQF > > Moreover, in case of a bridged call, could you add the possibility to > pass as is from header for an anonymous call. > Running current, I have this: > on A-leg: > From: > "Anonymous";tag=c0a80101-fed0c > on B-leg: > From: "Anonymous" >;tag=tpjD57U2pyrFN > > and I'd like B-leg to match A-leg for anonymous call as stated in RFC, > anonymous at anonymous.invalid is the proposed way to handle anonymous > call, I'll add 50$ more for support of this last request. > > thanks for your reactivity. > regards, > rod > > Anthony Minessale wrote: > > Latest SVN: > > > > Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > > > > > > > Send RPID (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > > > > > > > Send P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > > > > > > > Send RPID with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > > > [ Show ? ] > > Anthony Minessale II > > > added > > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > > > Send RPID > (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > > > Send > P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > > > Send RPID > with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > > > > > Send RPID with chosen content and privacy flags (+ delimited, > none to > > clear all flags) > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/user at example.com > > > > > > > > > > [ Show ? ] > > Anthony Minessale II > > > added > > a comment - 10/Mar/09 07:59 AM Send no extra caller id info: > > {sip_cid_type=none}sofia/default/user at example.com > > > > Send RPID > (default) > > {sip_cid_type=rpid}sofia/default/user at example.com > > > > Send > P-XXX-Identity > > {sip_cid_type=pid}sofia/default/user at example.com > > > > Send RPID > with chosen content > > > {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user at example.com > > > > > > > > > > Also the privacy app on the inbound leg controls the remaining > > contents of the RPID and Privacy headers. > > > > > > On Tue, Mar 10, 2009 at 7:32 AM, kokoska rokoska > > > >> > wrote: > > > > > > rod napsal(a): > > > ... if we could strip the RPID, we could write a new one, > but It > > > could be very convenient to get access to the fields in this > > header for > > > manipulation. > > > > > > > Yes, rod, this is exactly why I update bounty to $100 :-) > > > > Thank you very much, rod, for support! > > > > Best regards, > > > > kokoska.rokoska > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shanwlin at gmail.com Tue Mar 10 18:17:20 2009 From: shanwlin at gmail.com (shawn lin) Date: Wed, 11 Mar 2009 09:17:20 +0800 Subject: [Freeswitch-users] Where does the FS keep the call-id list? Message-ID: <9fcf45ed0903101817t3565832te8b166693e1ddbaa@mail.gmail.com> Hi all, I am new to FreeSwitch, I have a question when using SIPp <-> FreeSwitch <-> SIPp. The SIPp UAC create many calls and the SIPp UAS does reveive them. I wonder if anyone can tell me, where does the FreeSwitch keep the call-ids? Is there a list to contain all the call-id? If there is a list, how can I found it? Best Regards! Shawn Lin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/ee15d70f/attachment-0002.html From mike at jerris.com Wed Mar 11 04:34:38 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Mar 2009 07:34:38 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> Message-ID: The continue works fine, it just hangs up befoer that due to: Mike On Mar 11, 2009, at 2:46 AM, mszlazak at aol.com wrote: > I have the following in my dialplan. > > Individually, each extension does what it's suppose to do when > dialing 1111. > However, if I place continue=true in the first extension then it > alone gets executed and the succeeding extension does not. > I thought condition=true would allow the extension afterward to > execute. > > My test hardware for this dialplan is a single PSTN line. > A call comes in that line and "myExtension" executes then hopefully > hangs up to free the line. > Afterward, I want "myExtension_Continued" to execute the .js > application and dial out that single PSTN line. > > I need help in getting this scenerio to work. > > Thanks. > > > > > > data="effective_caller_id_number=${caller_id_number}"/> > data="hangup_after_bridge=true"/> > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/c33d4357/attachment-0002.html From mike at jerris.com Wed Mar 11 04:36:47 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Mar 2009 07:36:47 -0400 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <49B77126.2030500@ewetel.de> References: <49B77126.2030500@ewetel.de> Message-ID: <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> If your actually playing back a file (that blocks) it should work fine. Ringback goes in the background so it doesn't wait for it to "finish" before it does the ringback Mike On Mar 11, 2009, at 4:07 AM, Helmut Kuper wrote: > Hello, > > I'm looking for a way to play a file exactly once in early media state > of a call and then do e.g. a 302 reply. When I try it via ring_ready > and > ringback The playback immediately stops, when I send a respond. > > regards > Helmut > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Wed Mar 11 05:22:43 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Mar 2009 13:22:43 +0100 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> References: <49B77126.2030500@ewetel.de> <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> Message-ID: <49B7AD13.3040903@ewetel.de> Hi Mike, Thank you! regards Helmut On 11.03.2009 12:36, Michael Jerris wrote: > If your actually playing back a file (that blocks) it should work > fine. Ringback goes in the background so it doesn't wait for it to > "finish" before it does the ringback From anthony.minessale at gmail.com Wed Mar 11 06:16:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Mar 2009 08:16:38 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B7775D.3030308@laposte.net> References: <49AFC1C3.9030603@laposte.net> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> Message-ID: <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> On Wed, Mar 11, 2009 at 3:33 AM, rod wrote: > thanks a lot Anthony, > > I consider the work is done and I did the paypal transfer of 200$. > > But I'll be happy if you could do some minor changes: > > Done....r12563 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/0c7211ba/attachment-0002.html From helmut.kuper at ewetel.de Wed Mar 11 06:19:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Mar 2009 14:19:38 +0100 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <49B7AD13.3040903@ewetel.de> References: <49B77126.2030500@ewetel.de> <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> <49B7AD13.3040903@ewetel.de> Message-ID: <49B7BA6A.6030108@ewetel.de> Hi again, it works good. Can I assign a sleep to playback somehow to avoid loose of phrases at the beginning? regards Helmut On 11.03.2009 13:22, Helmut Kuper wrote: > Hi Mike, > > Thank you! > > regards > Helmut > > > On 11.03.2009 12:36, Michael Jerris wrote: > >> If your actually playing back a file (that blocks) it should work >> fine. Ringback goes in the background so it doesn't wait for it to >> "finish" before it does the ringback >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From shahal at jajah.com Wed Mar 11 07:08:28 2009 From: shahal at jajah.com (Shahal Hazan) Date: Wed, 11 Mar 2009 16:08:28 +0200 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app during bridge between softphone ext 1000 and PSTN. Message-ID: Hi, After I call the external number successfully, I'm able to receive DTMF from the softphone but the PSTN's DTMF doesn't work. I am able to change the mute status during a conference from the same PSTN. I had a look at: http://wiki.freeswitch.org/wiki/Dtmf_troubleshooting but it didn't help. I have: on the Local_Extension and on my SIP_PROVIDER extension, both at sofia/default Any ideas? Shahal Hazan Mobile Team Jajah My blog: http://jajahdevblog.com/shahal www.Jajah.com M: +972-54-227-9567 This mail was sent via Mail-SeCure System. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/0647daf3/attachment-0002.html From kawarod at laposte.net Wed Mar 11 07:09:30 2009 From: kawarod at laposte.net (rod) Date: Wed, 11 Mar 2009 18:09:30 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> Message-ID: <49B7C61A.6070201@laposte.net> ok, almost perfect :p using this in dialplan: {sip_cid_type=rpid,origination_caller_id_number=000000${CALLER_NUMBER},origination_caller_id_name=test,originatio\ n_privacy=screen+hide_name+hide_number,sip_from_uri=sip:anonymous at anonymous.invalid}sofia/ss7/000000${destination_number}@${GW_IP} I have this on B leg From: "test" ;tag=gQ53Hjp617BSH. Remote-Party-ID: "test" ;party=calling;screen=yes;privacy=off. As you can see the caller id name is still modified, do you think you could add something like sip_from_uri_name to circumvent this. Anthony, I tried many settings for origination_privacy and it seems to do nothing on the RPID header. Any clue? regards. Anthony Minessale wrote: > > > On Wed, Mar 11, 2009 at 3:33 AM, rod > wrote: > > thanks a lot Anthony, > > I consider the work is done and I did the paypal transfer of 200$. > > But I'll be happy if you could do some minor changes: > > > Done....r12563 > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From miles.chet at gmail.com Wed Mar 11 07:15:57 2009 From: miles.chet at gmail.com (roberto) Date: Wed, 11 Mar 2009 11:15:57 -0300 Subject: [Freeswitch-users] suggestions to application Message-ID: Hello, I have an app write in php running with asterisk for calling cards, i would like to migrate everything to freeswitch, but i think that fs has almost everything that a calling card need read to use.. I?m looking for suggestion on some features like: Accounting Rate Billing Route Please someone could give me any suggestion on wich program that i should use. Thanks, -- "Without love, we are birds with broken wings." Morrie From anthony.minessale at gmail.com Wed Mar 11 08:14:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Mar 2009 10:14:36 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B7C61A.6070201@laposte.net> References: <49AFC1C3.9030603@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> Message-ID: <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> try now On Wed, Mar 11, 2009 at 9:09 AM, rod wrote: > ok, > > almost perfect :p > > using this in dialplan: > > {sip_cid_type=rpid,origination_caller_id_number=000000${CALLER_NUMBER},origination_caller_id_name=test,originatio\ > > n_privacy=screen+hide_name+hide_number,sip_from_uri=sip:anonymous at anonymous.invalid > }sofia/ss7/000000${destination_number}@${GW_IP} > > I have this on B leg > From: "test" ;tag=gQ53Hjp617BSH. > Remote-Party-ID: "test" > > >;party=calling;screen=yes;privacy=off. > > As you can see the caller id name is still modified, do you think you > could add something like sip_from_uri_name to circumvent this. > > Anthony, I tried many settings for origination_privacy and it seems to > do nothing on the RPID header. Any clue? > > regards. > > Anthony Minessale wrote: > > > > > > On Wed, Mar 11, 2009 at 3:33 AM, rod > > wrote: > > > > thanks a lot Anthony, > > > > I consider the work is done and I did the paypal transfer of 200$. > > > > But I'll be happy if you could do some minor changes: > > > > > > Done....r12563 > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/a24d4979/attachment-0002.html From anthony.minessale at gmail.com Wed Mar 11 08:15:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Mar 2009 10:15:12 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> Message-ID: <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> oops hit send too soon, try now using sip_from_display variable to control the display field in the from url On Wed, Mar 11, 2009 at 10:14 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try now > > > On Wed, Mar 11, 2009 at 9:09 AM, rod wrote: > >> ok, >> >> almost perfect :p >> >> using this in dialplan: >> >> {sip_cid_type=rpid,origination_caller_id_number=000000${CALLER_NUMBER},origination_caller_id_name=test,originatio\ >> >> n_privacy=screen+hide_name+hide_number,sip_from_uri=sip:anonymous at anonymous.invalid >> }sofia/ss7/000000${destination_number}@${GW_IP} >> >> I have this on B leg >> From: "test" ;tag=gQ53Hjp617BSH. >> Remote-Party-ID: "test" >> >> >;party=calling;screen=yes;privacy=off. >> >> As you can see the caller id name is still modified, do you think you >> could add something like sip_from_uri_name to circumvent this. >> >> Anthony, I tried many settings for origination_privacy and it seems to >> do nothing on the RPID header. Any clue? >> >> regards. >> >> Anthony Minessale wrote: >> > >> > >> > On Wed, Mar 11, 2009 at 3:33 AM, rod > > > wrote: >> > >> > thanks a lot Anthony, >> > >> > I consider the work is done and I did the paypal transfer of 200$. >> > >> > But I'll be happy if you could do some minor changes: >> > >> > >> > Done....r12563 >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > >> > 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:213-799-1400 >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/40ee6a4c/attachment-0002.html From chris at maxpowersoft.com Wed Mar 11 08:25:57 2009 From: chris at maxpowersoft.com (Chris Danielson) Date: Wed, 11 Mar 2009 08:25:57 -0700 Subject: [Freeswitch-users] C API session and channels questions In-Reply-To: References: Message-ID: <49B7D805.50303@maxpowersoft.com> Hello Guys, I have a question regarding how I can do the following within the FS C API. What API can I use to list the current channels and sessions? What I have already in place is a generic FS module capable of serializing any sort of C data and storing it onto another device or caching server. I then have the ability to deserialize and operate on these generic objects already in my module. What I anticipate doing is finding a path to load up the FreeSWITCH (session and channel) objects and deserialize them into a freshly started FreeSWITCH instance. Any ideas on what switch_core_session.c methods, etc. I should use and any pitfalls? My primary focus for my company is conference calling, but I'd like to make this a means to work with everything that I possibly can. As soon as I get my basic testing and instances working I'll offer up the code to bkw, mikej and anthm for review. As-is, I'm not even sure if what I'm asking is even feasible, but I think having a module like this would be a good start to fill my needs and perhaps those of the community. So again, my two questions are as follows: 1) What API can I use to enumerate through the current channels and sessions initialized and running in memory? (I'm looking to clone and serialize them and store them on a separate server). 2) Any helper methods or ways that I can re-construct the channels and sessions into memory on a freshly started instance of FreeSWITCH? Kind Regards, Chris Danielson (aka. danchris) From msc at freeswitch.org Wed Mar 11 08:40:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 08:40:30 -0700 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <49B7BA6A.6030108@ewetel.de> References: <49B77126.2030500@ewetel.de> <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> <49B7AD13.3040903@ewetel.de> <49B7BA6A.6030108@ewetel.de> Message-ID: <87f2f3b90903110840p5ba9412dk8211377a4992fd3e@mail.gmail.com> > it works good. Can I assign a sleep to playback somehow to avoid loose > of phrases at the beginning? Are you playing just a single file? You can use the phrase macros to create a pause at the beginning of your playback. Or you can cheat and prepend a few hundred (or thousand) milliseconds of silence at the beginning of your sound file. :) -MC From msc at freeswitch.org Wed Mar 11 08:46:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 08:46:17 -0700 Subject: [Freeswitch-users] suggestions to application In-Reply-To: References: Message-ID: <87f2f3b90903110846j69d73a0ct58a4b41f3c06168e@mail.gmail.com> > I?m looking for suggestion on some features like: > > Accounting > Rate > Billing > Route There's nothing out-of-box ready, but you should look at these mods: mod_nibblebill - real-time billing mod_lcr - outbound routing mod_easyroute - inbound routing Start with the wiki: http:wiki.freeswitch.org There's a lot of research for you to do. Have fun! :) -MC From msc at freeswitch.org Wed Mar 11 08:51:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 08:51:06 -0700 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app during bridge between softphone ext 1000 and PSTN. In-Reply-To: References: Message-ID: <87f2f3b90903110851q1f1589bfq6102cf4c65c55130@mail.gmail.com> > After I call the external number successfully, I?m able to receive DTMF from > the softphone but the PSTN?s DTMF doesn?t work. We definitely don't want to assume anything, so I have to ask the obvious questions: who is the provider? are the DTMFs in-band or RFC2833? Any chance you can turn on full debugging and see if there are any clues? Thanks! -MC From msc at freeswitch.org Wed Mar 11 08:54:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 08:54:43 -0700 Subject: [Freeswitch-users] C API session and channels questions In-Reply-To: <49B7D805.50303@maxpowersoft.com> References: <49B7D805.50303@maxpowersoft.com> Message-ID: <87f2f3b90903110854x580eea6dkd42efab02c4c3ff8@mail.gmail.com> On Wed, Mar 11, 2009 at 8:25 AM, Chris Danielson wrote: > Hello Guys, > I have a question regarding how I can do the following within the FS C API. > The devs love it when people get down and dirty with FS! However, this thread is definitely better suited for the freeswitch-dev mailing list. Also, I notice that you're in IRC #freeswitch but not #freeswitch-dev. Hop in to the dev channel and you'll have a handful of people who will be willing to kick this stuff around with you. -MC From chris at maxpowersoft.com Wed Mar 11 09:07:26 2009 From: chris at maxpowersoft.com (Chris Danielson) Date: Wed, 11 Mar 2009 09:07:26 -0700 Subject: [Freeswitch-users] C API session and channels questions In-Reply-To: <87f2f3b90903110854x580eea6dkd42efab02c4c3ff8@mail.gmail.com> References: <49B7D805.50303@maxpowersoft.com> <87f2f3b90903110854x580eea6dkd42efab02c4c3ff8@mail.gmail.com> Message-ID: <49B7E1BE.8010504@maxpowersoft.com> Michael, Thanks! I'm joining up now. Regards, Chris Michael Collins wrote: > On Wed, Mar 11, 2009 at 8:25 AM, Chris Danielson wrote: > >> Hello Guys, >> I have a question regarding how I can do the following within the FS C API. >> >> > > The devs love it when people get down and dirty with FS! However, this > thread is definitely better suited for the freeswitch-dev mailing > list. Also, I notice that you're in IRC #freeswitch but not > #freeswitch-dev. Hop in to the dev channel and you'll have a handful > of people who will be willing to kick this stuff around with you. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/77a0ad64/attachment-0002.html From helmut.kuper at ewetel.de Wed Mar 11 09:31:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Mar 2009 17:31:38 +0100 Subject: [Freeswitch-users] Waiting for playing a file as early media In-Reply-To: <87f2f3b90903110840p5ba9412dk8211377a4992fd3e@mail.gmail.com> References: <49B77126.2030500@ewetel.de> <8D67570E-957F-44B9-90F3-072352794C39@jerris.com> <49B7AD13.3040903@ewetel.de> <49B7BA6A.6030108@ewetel.de> <87f2f3b90903110840p5ba9412dk8211377a4992fd3e@mail.gmail.com> Message-ID: <49B7E76A.5030901@ewetel.de> Hi Mike, yes, it's just one single file. I had the same idea to cheat a bit, but a dynamic solution is better ... will test the phrase macros. thx for the hint! regards helmut On 11.03.2009 16:40, Michael Collins wrote: >> it works good. Can I assign a sleep to playback somehow to avoid loose >> of phrases at the beginning? >> > Are you playing just a single file? You can use the phrase macros to > create a pause at the beginning of your playback. Or you can cheat and > prepend a few hundred (or thousand) milliseconds of silence at the > beginning of your sound file. :) > __ From kristian.kielhofner at gmail.com Wed Mar 11 09:32:34 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 12:32:34 -0400 Subject: [Freeswitch-users] STUN error In-Reply-To: <694251.83720.qm@web53610.mail.re2.yahoo.com> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> Message-ID: <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> On Mon, Mar 9, 2009 at 11:09 PM, Will Smith wrote: > Thank you Brian, it works like a champ. > > Yes,?west philadelfia born and raised? On a playground is where I spent most of my days... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From sprice at gmail.com Wed Mar 11 09:39:15 2009 From: sprice at gmail.com (SP) Date: Wed, 11 Mar 2009 11:39:15 -0500 Subject: [Freeswitch-users] STUN error In-Reply-To: <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> Message-ID: <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> On Wed, Mar 11, 2009 at 11:32, Kristian Kielhofner wrote: > On Mon, Mar 9, 2009 at 11:09 PM, Will Smith wrote: >> Thank you Brian, it works like a champ. >> >> Yes,?west philadelfia born and raised? > > On a playground is where I spent most of my days... Chilling out, maxing, relaxing all cool > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From mszlazak at aol.com Wed Mar 11 09:42:44 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 11 Mar 2009 12:42:44 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> Message-ID: <8CB707F89214902-624-883@webmail-da08.sysops.aol.com> I changed that tag so that hangup_after_bridge is false: ? but I still don't get the .js application working which is nothing more than a test script that ran if I dialed it's extension with the preceding one commented out: s = new Session("{ignore_early_media=true}sofia/gateway/spa3102/12223334444 at 10.0.0.5:5061"); while (s.ready()) { ??? s.answer(); ??? s.speak("cepstral","Callie","Hello World"); } -----Original Message----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Wed, 11 Mar 2009 4:34 am Subject: Re: [Freeswitch-users] Problem with "continue" in extension. The continue works fine, it just hangs up befoer that due to: ?? ? ? ? ? ? Mike On Mar 11, 2009, at 2:46 AM, mszlazak at aol.com wrote: I have the following in my dialplan. Individually, each extension does what it's suppose to do when dialing 1111. However, if I place continue=true in the first extension then it alone gets executed and the succeeding extension does not. I thought condition=true would allow the extension afterward to execute. My test hardware for this dialplan is a single PSTN line. A call comes in that line and "myExtension" executes then hopefully hangs up to free the line. Afterward, I want "myExtension_Continued" to execute the .js application and dial out that single PSTN line. I need help in getting this scenerio to work. Thanks. ?? ? ?? ??? ? ?? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ??? ? ?? ??? ? ?? ? ?? ??? ? ?? ??? ? ?? ??? ??? ? ?? ??? ? ?? ? = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/5c898ba0/attachment-0002.html From msc at freeswitch.org Wed Mar 11 10:14:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 10:14:08 -0700 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <8CB707F89214902-624-883@webmail-da08.sysops.aol.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> <8CB707F89214902-624-883@webmail-da08.sysops.aol.com> Message-ID: <87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com> On Wed, Mar 11, 2009 at 9:42 AM, wrote: > I changed that tag so that hangup_after_bridge is false: > > ? > but I still don't get the .js application working which is nothing more than > a test script that ran if I dialed it's extension with the preceding one > commented out: Try adding a break="never" to your first extension: I believe that will cause the the dialplan to keep looking for "1111" even after it has been matched once. -MC From benke at inqnet.at Wed Mar 11 10:07:42 2009 From: benke at inqnet.at (Christian Benke) Date: Wed, 11 Mar 2009 18:07:42 +0100 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" Message-ID: <20090311180742.0c6693cb@plex> Hi! I've recently started to configure a freeswitch for our new office pbx and so far i like it very much(Coming from asterisk&openser with 2 years experience at a ITSP. Openser was nice but i didn't like asterisk for several reasons, so i searched for a more stable and cleaner alternative. Freeswitch looks _very_ promising and i'd wished i could use it for more difficult demands than a simple office-pbx ;-)). So far i had little trouble(Though our installation doesn't require much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. The only issue i have not resolved yet is setting the outgoing DID("head"-number + extension, e.g. +4312345678 + 100). The relevant part of the default.xml looks like this atm(where +4312345678 is our "head"-phone-number without the extensions, ${caller_id_number} is a 3-digit extension, e.g.: 100): I'd expect with this dialplan the effective_caller_id would be in the "From:"-section of the INVITE, but it seems after the bridge it is overwritten with the gateway-username i've defined in the gateway-configuration in sip_profiles/external/. So instead of: From: "Desk Phone" ;tag=U6yQUSta2c2Xg. i get: From: "Desk Phone" ;tag=U6yQUSta2c2Xg. in the INVITE towards the sip-trunk. I may not have grasped yet how proper debugging with freeswitch works, however, in the console the last action i see, before the bridge to sofia/external is created, is the setting of the effective-caller-id, as expected(Do you want to see the whole output?). I guess i don't necessarily need to register with the provider, as they have configured the trunk for my ip-adress and i have theirs in the ACL(inbound calls work flawless with the head-number+extension), so maybe the registration is the reason why freeswitch does that automatically? It's probably a little issue, but i don't have the overview yet to understand how this happens, maybe someone can point me to the right place? Cheers Christian From mszlazak at aol.com Wed Mar 11 10:42:29 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 11 Mar 2009 13:42:29 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com><8CB707F89214902-624-883@webmail-da08.sysops.aol.com> <87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com> Message-ID: <8CB7087E25E4398-624-C4E@webmail-da08.sysops.aol.com> Mike, no luck with that either. I still need to see this through but another related approach will be needed later so I'll ask now. Does FreeSwitch have some script or something to set up an "auto dialer." Basically, I want to be able the store some caller info then have FS automatically check to see if a "reminder" calls need to be sent out. Thanks.? -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 11 Mar 2009 10:14 am Subject: Re: [Freeswitch-users] Problem with "continue" in extension. On Wed, Mar 11, 2009 at 9:42 AM, wrote: > I changed that tag so that hangup_after_bridge is false: > > ? > but I still don't get the .js application working which is nothing more than > a test script that ran if I dialed it's extension with the preceding one > commented out: Try adding a break="never" to your first extension: I believe that will cause the the dialplan to keep looking for "1111" even after it has been matched once. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/30c04acf/attachment-0002.html From miles.chet at gmail.com Wed Mar 11 10:48:37 2009 From: miles.chet at gmail.com (roberto) Date: Wed, 11 Mar 2009 14:48:37 -0300 Subject: [Freeswitch-users] suggestions to application In-Reply-To: <87f2f3b90903110846j69d73a0ct58a4b41f3c06168e@mail.gmail.com> References: <87f2f3b90903110846j69d73a0ct58a4b41f3c06168e@mail.gmail.com> Message-ID: thanks On Wed, Mar 11, 2009 at 12:46 PM, Michael Collins wrote: >> I?m looking for suggestion on some features like: >> >> Accounting >> Rate >> Billing >> Route > > There's nothing out-of-box ready, but you should look at these mods: > mod_nibblebill - real-time billing > mod_lcr - outbound routing > mod_easyroute - inbound routing > > Start with the wiki: http:wiki.freeswitch.org > > There's a lot of research for you to do. Have fun! :) > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- "Without love, we are birds with broken wings." Morrie From kristian.kielhofner at gmail.com Wed Mar 11 11:48:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 14:48:50 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? Message-ID: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> Hello everyone, I have an issue where FS seems to get confused in the presence of multiple Record-Route headers. SIP capture here: http://admin.star2star.com/fs-sip.log I've never seen this with FS before but it appears to process the multiple Record-Route headers backwards, at least in this case. I want to verify: 1) These Record-Route headers are syntactically correct (looks good to me). 2) FS should, in fact, process Record-Route headers "top down" and built its Route: headers (and reply) accordingly. At first I thought the FS/Sofia SIP parser may have been getting confused because the Record-Route from my proxy (.186) does not have a port in the URI. I tried adding a Record-Route header with a port - no difference. This is currently running trunk rev 12218 but I'm about to update to 12571 to see what happens. Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From msc at freeswitch.org Wed Mar 11 11:53:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 11:53:18 -0700 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <8CB7087E25E4398-624-C4E@webmail-da08.sysops.aol.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com> <8CB707F89214902-624-883@webmail-da08.sysops.aol.com> <87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com> <8CB7087E25E4398-624-C4E@webmail-da08.sysops.aol.com> Message-ID: <87f2f3b90903111153h23c477fei62a4bfaccf66664f@mail.gmail.com> On Wed, Mar 11, 2009 at 10:42 AM, wrote: > Mike, no luck with that either. > You may have to get fancy and use the transfer app and create an extension like this: Then just call it from the regular 1111 extension: > I still need to see this through but another related approach will be needed > later so I'll ask now. > > Does FreeSwitch have some script or something to set up an "auto dialer." > Basically, I want to be able the store some caller info then have FS > automatically check to see if a "reminder" calls need to be sent out. > you do you have the sched_api family of API functions: http://wiki.freeswitch.org/wiki/Mod_commands#sched_api I don't have it wikified but there is also an unsched_api function so that you can cancel a future scheduled api. The API could be something like "originate user/1000 &bridge(foo/bar)" -MC From msc at freeswitch.org Wed Mar 11 11:54:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Mar 2009 11:54:38 -0700 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> Message-ID: <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> > ?This is currently running trunk rev 12218 but I'm about to update to > 12571 to see what happens. To quote Samuel L. Jackson in "Jurassic Park": Hold on to your butts! -MC From kristian.kielhofner at gmail.com Wed Mar 11 12:21:36 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 15:21:36 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> Message-ID: <2d9149cd0903111221p7039037qe468768b959ce3d9@mail.gmail.com> On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins wrote: >> ?This is currently running trunk rev 12218 but I'm about to update to >> 12571 to see what happens. > > To quote Samuel L. Jackson in "Jurassic Park": > Hold on to your butts! > > -MC Yeah, I know. It's just that it's an AstLinux machine and my build machine is REALLY slow... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From mszlazak at aol.com Wed Mar 11 12:26:59 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 11 Mar 2009 15:26:59 -0400 Subject: [Freeswitch-users] Problem with "continue" in extension. In-Reply-To: <87f2f3b90903111153h23c477fei62a4bfaccf66664f@mail.gmail.com> References: <8CB702C44AB4BF8-1720-1F05@WEBMAIL-MZ04.sysops.aol.com><8CB707F89214902-624-883@webmail-da08.sysops.aol.com><87f2f3b90903111014i4c69fedas63e05874a9a20d8d@mail.gmail.com><8CB7087E25E4398-624-C4E@webmail-da08.sysops.aol.com> <87f2f3b90903111153h23c477fei62a4bfaccf66664f@mail.gmail.com> Message-ID: <8CB70967B6A20B2-B48-363@webmail-da08.sysops.aol.com> Mike, can you give me a pointer to what needs cleaning up so I have an idea how to make one of these? Also, thanks a lot for the sched_api and unsched_api, that's terrific news! -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 11 Mar 2009 11:53 am Subject: Re: [Freeswitch-users] Problem with "continue" in extension. On Wed, Mar 11, 2009 at 10:42 AM, wrote: > Mike, no luck with that either. > You may have to get fancy and use the transfer app and create an extension like this: Then just call it from the regular 1111 extension: > I still need to see this through but another related approach will be needed > later so I'll ask now. > > Does FreeSwitch have some script or something to set up an "auto dialer." > Basically, I want to be able the store some caller info then have FS > automatically check to see if a "reminder" calls need to be sent out. > you do you have the sched_api family of API functions: http://wiki.freeswitch.org/wiki/Mod_commands#sched_api I don't have it wikified but there is also an unsched_api function so that you can cancel a future scheduled api. The API could be something like "originate user/1000 &bridge(foo/bar)" -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/905a84e6/attachment-0002.html From Prometheus001 at gmx.net Wed Mar 11 14:03:27 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 11 Mar 2009 22:03:27 +0100 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app during bridge between softphone ext 1000 and PSTN. In-Reply-To: References: Message-ID: <49B8271F.7050705@gmx.net> My experience is, it can depend on the PSTN provider. I had problems with Sonus systems (see your SIP Invite message), there is a workaround in the wiki, but it didn't help in all my cases. I also have tested the start_dtmf application for additonal dtmf inband detection which helped in some cases. I also tried which didn't help in my cases + blocks routing of DTMFs through freeswitch. Best regards Peter Shahal Hazan schrieb: > > Hi, > > After I call the external number successfully, I?m able to receive > DTMF from the softphone but the PSTN?s DTMF doesn?t work. > > I am able to change the mute status during a conference from the same > PSTN. > > I had a look at: http://wiki.freeswitch.org/wiki/Dtmf_troubleshooting > but it didn?t help. > > I have: > > > > on the Local_Extension and on my SIP_PROVIDER extension, both at > sofia/default > > Any ideas? > > Shahal Hazan > > Mobile Team > Jajah > > My blog: http://jajahdevblog.com/shahal > > www.Jajah.com > > M: +972-54-227-9567 > > > > This mail was sent via Mail-SeCure System. > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From leon at scarlet-internet.nl Wed Mar 11 14:59:19 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Wed, 11 Mar 2009 22:59:19 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com> <49B0D603.502@laposte.net> <49B13277.9090505@3c.co.uk> <49B4BBA1.4000109@laposte.net> <49B64869.6080007@laposte.net> <49B64F69.5070209@post.cz> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <87f2f3b90903101348i6b27f95ag3ba496892ef9cb70@mail.gmail.com> Message-ID: <0DA60EA4-0873-48E9-A263-A20D14BF5A19@scarlet-internet.nl> Hi, I didn't try this new functionality yet, but shouldn't http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type be in a different section ? It's not SDP manipulation, or is it ? regards, Leon On Mar 10, 2009, at 9:48 PM, Michael Collins wrote: > On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale > wrote: >> Latest SVN: >> >> Send no extra caller id info: >> {sip_cid_type=none}sofia/default/user at example.com >> >> Send RPID (default) >> {sip_cid_type=rpid}sofia/default/user at example.com >> >> Send P-XXX-Identity >> {sip_cid_type=pid}sofia/default/user at example.com >> >> Send RPID with chosen content >> {sip_cid_type >> = >> rpid >> ,origination_caller_id_name >> =test,origination_caller_id_number=1234,origination_privacy=screen >> +hide_name+hide_number}sofia/defaul tuser at example.com > > FYI, I added this info to the channel variables page: > http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Wed Mar 11 15:19:47 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 11 Mar 2009 18:19:47 -0400 Subject: [Freeswitch-users] STUN error In-Reply-To: <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> Message-ID: <49B83903.3090306@freeswitch.org> SP wrote: > On Wed, Mar 11, 2009 at 11:32, Kristian Kielhofner > wrote: > >> On Mon, Mar 9, 2009 at 11:09 PM, Will Smith wrote: >> >>> Thank you Brian, it works like a champ. >>> >>> Yes, west philadelfia born and raised? >>> >> On a playground is where I spent most of my days... >> > > Chilling out, maxing, relaxing all cool > > OK, couldn't resist "shootin' some B-Ball outside the school" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/7381c524/attachment-0002.html From intralanman at freeswitch.org Wed Mar 11 15:24:00 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 11 Mar 2009 18:24:00 -0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> Message-ID: <49B83A00.7000706@freeswitch.org> Anthony Minessale wrote: > try now using sip_from_display variable to control the display field > in the from url It would be awesome if someone from the community would document all of these nice new variables on the wiki. -Ray From kjv at ken-ton.com Wed Mar 11 17:19:50 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Wed, 11 Mar 2009 20:19:50 -0400 Subject: [Freeswitch-users] STUN error In-Reply-To: <49B83903.3090306@freeswitch.org> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> <49B83903.3090306@freeswitch.org> Message-ID: <1AEFE3D0-F580-40F0-81E8-BA51E4385BD5@ken-ton.com> Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-448-3009 x0 On Mar 11, 2009, at 6:19 PM, Raymond Chandler wrote: > SP wrote: >> >> On Wed, Mar 11, 2009 at 11:32, Kristian Kielhofner >> wrote: >> >>> On Mon, Mar 9, 2009 at 11:09 PM, Will Smith >>> wrote: >>> >>>> Thank you Brian, it works like a champ. >>>> >>>> Yes, west philadelfia born and raised? >>>> >>> On a playground is where I spent most of my days... >>> >> Chilling out, maxing, relaxing all cool >> >> > OK, couldn't resist > "shootin' some B-Ball outside the school" When a couple a' guys who were up to no good > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/16e2135c/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/16e2135c/attachment-0002.bin From brian at freeswitch.org Wed Mar 11 17:28:51 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Mar 2009 19:28:51 -0500 Subject: [Freeswitch-users] STUN error In-Reply-To: <1AEFE3D0-F580-40F0-81E8-BA51E4385BD5@ken-ton.com> References: <61E33987-F7B7-4A7E-9C12-BE9351279296@freeswitch.org> <694251.83720.qm@web53610.mail.re2.yahoo.com> <2d9149cd0903110932s2f059777jc0999d28343d4e99@mail.gmail.com> <7e2ac3270903110939p6414593cg15f601cda48b2711@mail.gmail.com> <49B83903.3090306@freeswitch.org> <1AEFE3D0-F580-40F0-81E8-BA51E4385BD5@ken-ton.com> Message-ID: <80C0FDB7-83D7-40D7-9090-C24D5022FF6C@freeswitch.org> On Mar 11, 2009, at 7:19 PM, Karl Vesterling wrote: >> OK, couldn't resist >> "shootin' some B-Ball outside the school" > When a couple a' guys who were up to no good >> Started making trouble in my neighbourhood I got in one little fight and my mom got scared And said youre moving with your aunte and uncle in bel-air /b PS: fun has been had! From kristian.kielhofner at gmail.com Wed Mar 11 19:54:46 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 22:54:46 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> Message-ID: <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins wrote: >> ?This is currently running trunk rev 12218 but I'm about to update to >> 12571 to see what happens. > > To quote Samuel L. Jackson in "Jurassic Park": > Hold on to your butts! > > -MC Just got around to trying again on 12571 - same result. Here it is again just the OK and the ACK this time: U 208.38.149.186:5060 -> 71.228.78.51:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 71.228.78.51:5060;received=71.228.78.51;rport=5060;branch=z9hG4bK4D194FH359yaH. To: ;tag=as4ba5ae74. From: "Extension 1000" ;tag=4veH7cg0XS04r. Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. CSeq: 112294452 INVITE. Content-Type: application/sdp. Contact: . Content-Length: 285. Record-Route: . User-Agent: Packetrino. Supported: replaces. Record-Route: . Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. . v=0. o=root 10960 10961 IN IP4 64.2.142.73. s=session. c=IN IP4 64.2.142.73. t=0 0. m=audio 18680 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/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 - - - -. a=ptime:20. a=sendrecv. U 71.228.78.51:5060 -> 64.2.142.93:5060 ACK sip:19412848354 at 64.2.142.73 SIP/2.0. Via: SIP/2.0/UDP 71.228.78.51;rport;branch=z9hG4bK5pt26a262jNXc. Route: . Route: . Max-Forwards: 70. From: "Extension 1000" ;tag=4veH7cg0XS04r. To: ;tag=as4ba5ae74. Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. CSeq: 112294452 ACK. Contact: . Content-Length: 0. . Trying to be as self-sufficient as I can, it looks like the code for this is on line 4268 of src/mod/endpoints/mod_sofia/sofia.c. I just wish I new what to do to it... ;) Am I the only one that has experienced this? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Wed Mar 11 20:13:13 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 23:13:13 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> Message-ID: <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> After reading into this more it looks like the Record-Route headers are to be parsed in reverse order (which FS is doing). Sorry! On Wed, Mar 11, 2009 at 10:54 PM, Kristian Kielhofner wrote: > On Wed, Mar 11, 2009 at 2:54 PM, Michael Collins wrote: >>> ?This is currently running trunk rev 12218 but I'm about to update to >>> 12571 to see what happens. >> >> To quote Samuel L. Jackson in "Jurassic Park": >> Hold on to your butts! >> >> -MC > > Just got around to trying again on 12571 - same result. ?Here it is > again just the OK and the ACK this time: > > U 208.38.149.186:5060 -> 71.228.78.51:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > 71.228.78.51:5060;received=71.228.78.51;rport=5060;branch=z9hG4bK4D194FH359yaH. > To: ;tag=as4ba5ae74. > From: "Extension 1000" ;tag=4veH7cg0XS04r. > Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. > CSeq: 112294452 INVITE. > Content-Type: application/sdp. > Contact: . > Content-Length: 285. > Record-Route: . > User-Agent: Packetrino. > Supported: replaces. > Record-Route: . > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > . > v=0. > o=root 10960 10961 IN IP4 64.2.142.73. > s=session. > c=IN IP4 64.2.142.73. > t=0 0. > m=audio 18680 RTP/AVP 0 18 101. > a=rtpmap:0 PCMU/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 - - - -. > a=ptime:20. > a=sendrecv. > > > U 71.228.78.51:5060 -> 64.2.142.93:5060 > ACK sip:19412848354 at 64.2.142.73 SIP/2.0. > Via: SIP/2.0/UDP 71.228.78.51;rport;branch=z9hG4bK5pt26a262jNXc. > Route: . > Route: . > Max-Forwards: 70. > From: "Extension 1000" ;tag=4veH7cg0XS04r. > To: ;tag=as4ba5ae74. > Call-ID: dec5a4c0-8951-122c-a78f-f96fa82849d1. > CSeq: 112294452 ACK. > Contact: . > Content-Length: 0. > . > > ?Trying to be as self-sufficient as I can, it looks like the code for > this is on line 4268 of src/mod/endpoints/mod_sofia/sofia.c. ?I just > wish I new what to do to it... ;) > > ?Am I the only one that has experienced this? > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Wed Mar 11 20:18:47 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Mar 2009 22:18:47 -0500 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> Message-ID: Its ok... at the very least you question the norm! ;) /b On Mar 11, 2009, at 10:13 PM, Kristian Kielhofner wrote: > After reading into this more it looks like the Record-Route headers > are to be parsed in reverse order (which FS is doing). > > Sorry! From kristian.kielhofner at gmail.com Wed Mar 11 20:29:04 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 11 Mar 2009 23:29:04 -0400 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> Message-ID: <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> Aww, thanks Brian! On Wed, Mar 11, 2009 at 11:18 PM, Brian West wrote: > Its ok... at the very least you question the norm! ;) > > /b > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Wed Mar 11 20:34:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Mar 2009 22:34:36 -0500 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> Message-ID: ;) I expect to see you at cluecon this year? /b On Mar 11, 2009, at 10:29 PM, Kristian Kielhofner wrote: > Aww, thanks Brian! > > On Wed, Mar 11, 2009 at 11:18 PM, Brian West > wrote: >> Its ok... at the very least you question the norm! ;) >> >> /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/2b265fe2/attachment-0002.html From msc at freeswitch.org Wed Mar 11 20:50:13 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 11 Mar 2009 20:50:13 -0700 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> Message-ID: <83CD0007-CCC2-4420-95C5-8EC26D5C7FD2@freeswitch.org> On Mar 11, 2009, at 8:34 PM, Brian West wrote: > ;) I expect to see you at cluecon this year? > > /b > Notice how he threw in a compliment and an invite to CC but didn't actually address the question? ;) pretty sneaky bkw! BTW, the best way to come to CC is to get your boss to sponsor the event! :p -MC From brian at freeswitch.org Wed Mar 11 20:54:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Mar 2009 22:54:03 -0500 Subject: [Freeswitch-users] FS Confusion with multiple SIP Record-Route headers? In-Reply-To: <83CD0007-CCC2-4420-95C5-8EC26D5C7FD2@freeswitch.org> References: <2d9149cd0903111148l5d4e081ehd3f1ed843f35d8ef@mail.gmail.com> <87f2f3b90903111154y484616cdg30a4eb3f2b66c4ee@mail.gmail.com> <2d9149cd0903111954w1a9792d6y32a8e2fc2f557de0@mail.gmail.com> <2d9149cd0903112013t607b6f21s94a8216ac4ffa68d@mail.gmail.com> <2d9149cd0903112029q124dcbe0ia9db6160ac27839a@mail.gmail.com> <83CD0007-CCC2-4420-95C5-8EC26D5C7FD2@freeswitch.org> Message-ID: <1C4F8A51-66B9-4C7D-B187-996521A3AEB6@freeswitch.org> I'm like bugs bunny here... sneaky wabbit! /b On Mar 11, 2009, at 10:50 PM, Michael S Collins wrote: > On Mar 11, 2009, at 8:34 PM, Brian West wrote: > >> ;) I expect to see you at cluecon this year? >> >> /b >> > Notice how he threw in a compliment and an invite to CC but didn't > actually address the question? ;) pretty sneaky bkw! > > BTW, the best way to come to CC is to get your boss to sponsor the > event! :p > > -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090311/d677e88f/attachment-0002.html From wiltingtree at gmail.com Wed Mar 11 22:02:46 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Thu, 12 Mar 2009 01:02:46 -0400 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails Message-ID: Hi, I'm trying to use FreeSWITCH's mod_voicemail to replace the voicemail on wireless phones. Does anybody know how to make the wireless phone know there is a voicemail waiting, so it can notify the user?Thanks for the help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/9dea6725/attachment-0002.html From krice at freeswitch.org Wed Mar 11 22:09:35 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 12 Mar 2009 00:09:35 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: Message-ID: When you say wireless do you mean like Cellular Phone? From: Adam Wilt Reply-To: Date: Thu, 12 Mar 2009 01:02:46 -0400 To: Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails Hi, I'm trying to use FreeSWITCH's mod_voicemail to replace the voicemail on wireless phones. Does anybody know how to make the wireless phone know there is a voicemail waiting, so it can notify the user? Thanks for the help! _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/70ad5e87/attachment-0002.html From kawarod at laposte.net Thu Mar 12 03:15:08 2009 From: kawarod at laposte.net (rod) Date: Thu, 12 Mar 2009 14:15:08 +0400 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> References: <49AFC1C3.9030603@laposte.net> <49B6595E.6040700@laposte.net> <49B65DF3.9060001@post.cz> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> Message-ID: <49B8E0AC.6070307@laposte.net> Thanks a lot Anthony, it's working great. I'm just checking the origination_privacy parameter, cause it seems to do nothing in my setup. Anthony Minessale wrote: > oops hit send too soon, > > try now using sip_from_display variable to control the display field > in the from url > > > On Wed, Mar 11, 2009 at 10:14 AM, Anthony Minessale > > wrote: > > try now > > > On Wed, Mar 11, 2009 at 9:09 AM, rod > wrote: > > ok, > > almost perfect :p > > using this in dialplan: > {sip_cid_type=rpid,origination_caller_id_number=000000${CALLER_NUMBER},origination_caller_id_name=test,originatio\ > n_privacy=screen+hide_name+hide_number,sip_from_uri=sip:anonymous at anonymous.invalid}sofia/ss7/000000${destination_number}@${GW_IP} > > I have this on B leg > From: "test" ;tag=gQ53Hjp617BSH. > Remote-Party-ID: "test" > >;party=calling;screen=yes;privacy=off. > > As you can see the caller id name is still modified, do you > think you > could add something like sip_from_uri_name to circumvent this. > > Anthony, I tried many settings for origination_privacy and it > seems to > do nothing on the RPID header. Any clue? > > regards. > > Anthony Minessale wrote: > > > > > > On Wed, Mar 11, 2009 at 3:33 AM, rod > > >> > wrote: > > > > thanks a lot Anthony, > > > > I consider the work is done and I did the paypal > transfer of 200$. > > > > But I'll be happy if you could do some minor changes: > > > > > > Done....r12563 > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shahal at jajah.com Thu Mar 12 03:41:30 2009 From: shahal at jajah.com (Shahal Hazan) Date: Thu, 12 Mar 2009 12:41:30 +0200 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 48 In-Reply-To: References: Message-ID: > After I call the external number successfully, I?m able to receive DTMF from > the softphone but the PSTN?s DTMF doesn?t work. We definitely don't want to assume anything, so I have to ask the obvious questions: who is the provider? are the DTMFs in-band or RFC2833? Any chance you can turn on full debugging and see if there are any clues? Thanks! -MC Hi, After turning on the debug in CLI I typed *3 on the PSTN and I got: 2009-03-12 11:22:49 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(0) 2009-03-12 11:22:51 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(3) Please note that I got DTMF(0) and not a DTMF(*)! I also double checked with wireshark and saw that the DTMF is SIP based, and the values were *3 and not 03 as FreeSWITCH reports. This probably is the problem and not the bind_meta_app. We are using bezeq international as our provider. DTMF are RFC2833. Thanks, Shahal This mail was sent via Mail-SeCure System. From shahal at jajah.com Thu Mar 12 03:43:33 2009 From: shahal at jajah.com (Shahal Hazan) Date: Thu, 12 Mar 2009 12:43:33 +0200 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app In-Reply-To: References: Message-ID: > After I call the external number successfully, I?m able to receive DTMF from > the softphone but the PSTN?s DTMF doesn?t work. We definitely don't want to assume anything, so I have to ask the obvious questions: who is the provider? are the DTMFs in-band or RFC2833? Any chance you can turn on full debugging and see if there are any clues? Thanks! -MC Hi, After turning on the debug in CLI I typed *3 on the PSTN and I got: 2009-03-12 11:22:49 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(0) 2009-03-12 11:22:51 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() INFO DTMF(3) Please note that I got DTMF(0) and not a DTMF(*)! I also double checked with wireshark and saw that the DTMF is SIP based, and the values were *3 and not 03 as FreeSWITCH reports. This probably is the problem and not the bind_meta_app. We are using bezeq international as our provider. DTMF are RFC2833. Thanks, Shahal This mail was sent via Mail-SeCure System. From shahal at jajah.com Thu Mar 12 03:43:53 2009 From: shahal at jajah.com (Shahal Hazan) Date: Thu, 12 Mar 2009 12:43:53 +0200 Subject: [Freeswitch-users] Recall: Freeswitch-users Digest, Vol 33, Issue 48 Message-ID: Shahal Hazan would like to recall the message, "Freeswitch-users Digest, Vol 33, Issue 48". This mail was sent via Mail-SeCure System. From william.suffill at gmail.com Thu Mar 12 05:36:27 2009 From: william.suffill at gmail.com (William Suffill) Date: Thu, 12 Mar 2009 08:36:27 -0400 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: References: Message-ID: <6b65470d0903120536m44a431aawaad1241172692358@mail.gmail.com> If they aren't attached directly to the FS box I guess you could call them and play a recording and allow them to access their VM by creating an outbound call w/ an ivr to a) see if u got the person vs VM etc then b) allow them to get to their VM. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/4c29c29d/attachment-0002.html From yudha2008 at gmail.com Thu Mar 12 06:47:44 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 12 Mar 2009 19:17:44 +0530 Subject: [Freeswitch-users] openzap with A102 Message-ID: Hi, I am using sangoma A102 card with freeswitch. I have updated all the changes in the freeswitch and have loaded openzap also. But still i cant able to make an outbound call. *openzap.conf* [span wanpipe] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 [span wanpipe] name => OpenZAP number => 2 trunk_type => E1 b-channel => 2:1-15 d-channel => 2:16 b-channel => 2:17-31 *openzap.conf.xml* *default.xml * - * I have attached the freeswitch log http://pastebin.freeswitch.org/7730 Can any one correct were i am wrong. * -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/95cc8b7b/attachment-0002.html From kawarod at laposte.net Thu Mar 12 07:12:04 2009 From: kawarod at laposte.net (rod) Date: Thu, 12 Mar 2009 18:12:04 +0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value Message-ID: <49B91834.6050004@laposte.net> Hi list, I'm testing mod_limit like this: where AREA could be any value like: US/EUROPE/AFRICA... that has been set for the call then I use sipp for load testing with 4 cps and max 65 calls so that I will be limited by the limit of 60 in the dialplan. I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit value growing up to 60. Sipp still tries to establish new call (up to 65 calls) at 4cps and for each new cps in excess, FS sends a 503. I wait for 10 seconds and stop sipp, but the limit value is never decreasing even when there is no more channels used (show channels), the limit value is stuck to 60. If I limit Sipp to 55 calls (below the limit value), the limit value increase and decrease depending on load, and the pbm doesn't appear. Does anybody is using mod_limit and have encountered the same pbm. I'm using latest svn: 12580. regards, rod From mrene_lists at avgs.ca Thu Mar 12 08:04:24 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 12 Mar 2009 11:04:24 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49B91834.6050004@laposte.net> References: <49B91834.6050004@laposte.net> Message-ID: <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> Were you doing transfers (action="transfer" in the dialplan) ? If yes, retry with revision 12581, and open a JIRA if it still is an issue (also include a full debug log (delete your freeswitch.log file before doing your test and attach it after) in your JIRA) Math On 12-Mar-09, at 10:12 AM, rod wrote: > Hi list, > > I'm testing mod_limit like this: > > > > > where AREA could be any value like: US/EUROPE/AFRICA... that has been > set for the call > > then I use sipp for load testing with 4 cps and max 65 calls so that I > will be limited by the limit of 60 in the dialplan. > > I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit > value > growing up to 60. > > Sipp still tries to establish new call (up to 65 calls) at 4cps and > for > each new cps in excess, FS sends a 503. > I wait for 10 seconds and stop sipp, but the limit value is never > decreasing even when there is no more channels used (show channels), > the > limit value is stuck to 60. > > > If I limit Sipp to 55 calls (below the limit value), the limit value > increase and decrease depending on load, and the pbm doesn't appear. > > Does anybody is using mod_limit and have encountered the same pbm. > > I'm using latest svn: 12580. > > regards, > rod > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Mark.Tabron at rnid-typetalk.org.uk Thu Mar 12 06:54:49 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Thu, 12 Mar 2009 13:54:49 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 Message-ID: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> Hi, My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn't quite correct. Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel. The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out) Here are the confs / dialplan as I have them: [span wanpipe] name => OpenZAP number => 1 trunk_type => e1 b-channel => 1:1-9 d-channel => 1:16 Do the config files look ok? Is there anything else I should be configuring? Is there anything else I can use to debug or get information on the PRI status? Thanks in advance. Mark. Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/b111162a/attachment-0002.html From ax.russo at gmail.com Thu Mar 12 08:01:46 2009 From: ax.russo at gmail.com (Alessandro Russo) Date: Thu, 12 Mar 2009 16:01:46 +0100 Subject: [Freeswitch-users] Freeswitch and OPAL/H323 Message-ID: Hi to all, I am a newbie in Freeswitch (FS). I have already installed a FS machine, following the wiki installation procedure, and I have also added the opal module following this procedure: http://wiki.freeswitch.org/wiki/Mod_opal When I running FS ###################################### freeswitch at atest> module_exists mod_opal API CALL [module_exists(mod_opal)] output: true freeswitch at atest> ###################################### I think I'have installed it correctly. My goal is to provide a conference tool for incoming h323-calls that come from a cisco call manager: we are behind a Cisco VoIP cloud. Every time a PSTN phone calls the number 1234-123456 the call manager knows that the extension 3456 has to be redirected to 192.168.193.38, that is the IP address of the FS machine, where file dialplan/default/myconference.xml contains the following lines ###################################### ###################################### On the other hand, whenever a user of FS calls an local extension (like 1XXX ), what I want is that FS forward this call to the cisco call manager through opal/h323 therefore I have a file in ###################################### ###################################### but it fails: when a FS user calls 1500, FS returns this message ###################################### 2009-03-12 15:55:08 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1000 at 192.168.193.38[c7b69402-0f15-11de-b4dc-c11b39fce37c] 2009-03-12 15:55:08 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1500 in context default 2009-03-12 15:55:08 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel opal/h323:1500 at IP.CALL.MANA.GER:1720 [c7c010e0-0f15-11de-b4dc-c11b39fce37c] 2009-03-12 15:55:08 [INFO] h323pdu.cxx:999 H225() Read error (0): 2009-03-12 15:55:08 [NOTICE] mod_opal.cpp:591 OnReleased() Hangup opal/h323:1500 at IP.CALL.MANA.GER:1720 [CS_CONSUME_MEDIA] [UNKNOWN] 2009-03-12 15:55:08 [INFO] tlibthrd.cxx:363 PWLib() Destroyed thread 0xb171a708 H225 Caller:0xa9587b90(id = 0) 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 22 (opal/h323:1500 at IP.CALL.MANA.GER:1720) Ended 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel opal/h323:1500 at IP.CALL.MANA.GER:1720 [CS_HANGUP] 2009-03-12 15:55:08 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: UNKNOWN 2009-03-12 15:55:08 [NOTICE] mod_dptools.c:596 hangup_function() Hangup sofia/internal/1000 at 192.168.193.38 [CS_EXECUTE] [NORMAL_CLEARING] 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 21 (sofia/internal/1000 at 192.168.193.38) Ended 2009-03-12 15:55:08 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/ 1000 at 192.168.193.38 [CS_HANGUP] ###################################### I don't understand why.... Any suggestions... Thanks Alessandro R. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/09216e6d/attachment-0002.html From cstomi.levlist at gmail.com Thu Mar 12 09:06:57 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Thu, 12 Mar 2009 17:06:57 +0100 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> Message-ID: <49B93321.5080500@gmail.com> Hello, we had the same problem. we couldn't test r12581 for sure yet, but we will. This fix is only for the limit (db version), right? Would be limit_hash a better choice to increase performance anyway? Rod, do yoo have maybe experiences with limit_hash with your sipp? Thanks in advance, Tamas Mathieu Rene ?rta: > Were you doing transfers (action="transfer" in the dialplan) ? > > If yes, retry with revision 12581, and open a JIRA if it still is an > issue (also include a full debug log (delete your freeswitch.log file > before doing your test and attach it after) in your JIRA) > > Math > > On 12-Mar-09, at 10:12 AM, rod wrote: > > >> Hi list, >> >> I'm testing mod_limit like this: >> >> >> >> >> where AREA could be any value like: US/EUROPE/AFRICA... that has been >> set for the call >> >> then I use sipp for load testing with 4 cps and max 65 calls so that I >> will be limited by the limit of 60 in the dialplan. >> >> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >> value >> growing up to 60. >> >> Sipp still tries to establish new call (up to 65 calls) at 4cps and >> for >> each new cps in excess, FS sends a 503. >> I wait for 10 seconds and stop sipp, but the limit value is never >> decreasing even when there is no more channels used (show channels), >> the >> limit value is stuck to 60. >> >> >> If I limit Sipp to 55 calls (below the limit value), the limit value >> increase and decrease depending on load, and the pbm doesn't appear. >> >> Does anybody is using mod_limit and have encountered the same pbm. >> >> I'm using latest svn: 12580. >> >> regards, >> rod >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu Mar 12 09:50:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Mar 2009 09:50:07 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> > My first post to the list. I?m a bit of a newb to FreeSwitch (and linux) so > apologies if some of my terminology isn?t quite correct. Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > > > > Recently had a 9 channel ISDN30 (euro ? q931) installed by BT (UK). We?ve > hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the > card is green and wanrouter is installed and up in TDM_API mode, with the > connection status showing as connected. ?Configured Openzap for 9 b and 1 d > channel as described in Freeswitch Wiki. Then created a diaplan to fire off > any calls preceded by 9 to the next available openzap channel. Looks good so far... > The problem I have is when I initiate an external call (using 9xxxxxxx) from > an extension I can see Freeswitch allocating the call to the next available > channel but then the just sits there and times out after 1 minute. With the > cause stated as ORIGINATOR_CANCEL (guessing this is the time out) okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has lots of useful information for how to gather log information, how to use the pastebin, etc. Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers: To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org). Welcome to the wonderful world of telephony debugging! -MC P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.) From msc at freeswitch.org Thu Mar 12 09:57:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Mar 2009 09:57:38 -0700 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: References: Message-ID: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> > I have attached the freeswitch log http://pastebin.freeswitch.org/7730 > > Can any one correct were i am wrong. Baskar, At first look your configs seem okay. Please pastebin the output of these commands from the CLI: oz list oz dump 1 -MC From msc at freeswitch.org Thu Mar 12 10:04:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Mar 2009 10:04:53 -0700 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app In-Reply-To: References: Message-ID: <87f2f3b90903121004y1e4bc35epe3eef28da0993a65@mail.gmail.com> > I also double checked with wireshark and saw that the DTMF is SIP based, and the values were *3 and not 03 as FreeSWITCH reports. > This probably is the problem and not the bind_meta_app. > We are using bezeq international as our provider. > DTMF are RFC2833. > Thanks, > Shahal Shahal, The devs have very recently made some improvements. Can you check out the latest SVN and try this again? Let us know what happens. -MC From brian at freeswitch.org Thu Mar 12 10:13:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Mar 2009 12:13:22 -0500 Subject: [Freeswitch-users] No DTMF received from PSTN when using bind_meta_app In-Reply-To: References: Message-ID: <63832FB2-689F-4B8A-8896-ABCABD45E856@freeswitch.org> What SVN Rev are you on? I recall fixing this ... /b On Mar 12, 2009, at 5:43 AM, Shahal Hazan wrote: >> After I call the external number successfully, I?m able to receive >> DTMF from >> the softphone but the PSTN?s DTMF doesn?t work. > > We definitely don't want to assume anything, so I have to ask the > obvious questions: > who is the provider? > are the DTMFs in-band or RFC2833? > > Any chance you can turn on full debugging and see if there are any > clues? > > Thanks! > -MC > > Hi, > After turning on the debug in CLI I typed *3 on the PSTN and I got: > 2009-03-12 11:22:49 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() > INFO DTMF(0) > 2009-03-12 11:22:51 [DEBUG] sofia.c:3744 sofia_handle_sip_i_info() > INFO DTMF(3) > Please note that I got DTMF(0) and not a DTMF(*)! > I also double checked with wireshark and saw that the DTMF is SIP > based, and the values were *3 and not 03 as FreeSWITCH reports. > This probably is the problem and not the bind_meta_app. > We are using bezeq international as our provider. > DTMF are RFC2833. > Thanks, > Shahal > > > > This mail was sent via Mail-SeCure System. From e.schmidbauer at gmail.com Thu Mar 12 11:38:33 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Thu, 12 Mar 2009 14:38:33 -0400 Subject: [Freeswitch-users] RPC and web admin panel for conference? In-Reply-To: References: Message-ID: <2cef777b0903121138p1c55c7h8b4eb71427d0b901@mail.gmail.com> I would like to try out and possibly contribute to your web admin panel. I was wondering if i need flash 8 or Flash MX 2004 for this to work though. Please let me know because I do not have a copy of that software and I am not sure if I can run it on linux (which I am running freeswitch on). On Tue, Mar 10, 2009 at 6:32 PM, Ken Rice wrote: > Hey, I just implemented something like this and commited it to my contrib > directory (scripts/contrib/swk ) its a mixture of amf-php, ESL, and Flex... > Its not complete by anymeans and you need Flex3 to compile the UI... > > Anyone wanting to throw some patches at it for other functionality are > welcome to do so... One thing is severly lacks at this time is ANY sort of > authentication...so you wouldn?t want it publically open to the world > > K > > > ________________________________ > From: Harry FSwitch > Reply-To: > Date: Mon, 9 Mar 2009 18:17:41 -0400 > To: > Subject: [Freeswitch-users] RPC and web admin panel for conference? > > Hi all, > > I'm looking to implement an admin panel much like the one used at > http://conference.freeswitch.org. Now I obviously cannot login and see the > "admin" part of the panel but I'm pretty sure whats in there. > > I have xml_rpc running and can connect via http and issue commands. I've > searched the forum here and went through the wiki, found nothing that looked > like a panel. I was hoping to find a panel I can just configure and > implement. Does anyone have a php (I guess, seeing as I have a php server) > panel they can share with me? I'm sure I can get it working for my system. > The thought of attempting one on my own at THIS point seems daunting at > best. > > Any help would be greatly appreciated! > > Thanks > > ________________________________ > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From switchserver at gmail.com Thu Mar 12 11:55:10 2009 From: switchserver at gmail.com (Harry FSwitch) Date: Thu, 12 Mar 2009 14:55:10 -0400 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! Message-ID: Greetings, Last week I submitted this post to the mailing list... http://www.nabble.com/Please-end-the-torment-td22352222.html I received a mixed response to say the least, no one however emailed me saying they wanted to be involved or would contribute hosting resources. I figured eh screw-it, and went ahead and created the forums anyway and opened them today! http://freeswitch411.info I went on IRC to let folks know its available and after a few jabs from the crowd Anthony said: "#freeswitch 2009-03-12 10:47:04 [anthm] harr, if you maintain it you are welcome to have it" If it takes off and provides a friendly, helpful entryway for new FreeSWITCH users then I will be happy, if it flops I'll be said. But I will maintain it and do what I can to help FreeSWITCH grow. This is the only time I'll overtly mention it on this list, I'll have the link in my signature and thats about it. :) Thanks for your attention -- Harry http://freeswitch411.info -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/f8025f5e/attachment-0002.html From intralanman at freeswitch.org Thu Mar 12 12:00:44 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 12 Mar 2009 15:00:44 -0400 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! In-Reply-To: References: Message-ID: <49B95BDC.9030407@freeswitch.org> Harry FSwitch wrote: > > If it takes off and provides a friendly, helpful entryway for new > FreeSWITCH users then I will be happy, if it flops I'll be said. But I > will maintain it and do what I can to help FreeSWITCH grow. This is > the only time I'll overtly mention it on this list, I'll have the link > in my signature and thats about it. :) > something like the following? -------------- unofficial forums http://freeswitch411.info From krice at suspicious.org Thu Mar 12 12:20:21 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 12 Mar 2009 14:20:21 -0500 Subject: [Freeswitch-users] RPC and web admin panel for conference? In-Reply-To: <2cef777b0903121138p1c55c7h8b4eb71427d0b901@mail.gmail.com> Message-ID: Its a 2 part solution Part 1 is PHP + AMF-PHP + ESL AMF-PHP is a class for PHP to do the Flash/Flex Serialized Binary Data format for communication to the back end ESL is FS's Event Socket Library that has been wrapped for use in FreeSwitch The Flex Client requires at least Flash 9 runtime to run... To compile it, you'll need either the Flex IDE, the Eclipse Flex plugin (both from adobe) or the opensource flex compiler In its current form its not meant to be a 'user facing' controller, but it can be used as a starting point. I have tried to be generic as possible about what I am implementing. K > From: e schmidbauer > Reply-To: > Date: Thu, 12 Mar 2009 14:38:33 -0400 > To: > Subject: Re: [Freeswitch-users] RPC and web admin panel for conference? > > I would like to try out and possibly contribute to your web admin > panel. I was wondering if i need flash 8 or Flash MX 2004 for this to > work though. Please let me know because I do not have a copy of that > software and I am not sure if I can run it on linux (which I am > running freeswitch on). > > On Tue, Mar 10, 2009 at 6:32 PM, Ken Rice wrote: >> Hey, I just implemented something like this and commited it to my contrib >> directory (scripts/contrib/swk ) its a mixture of amf-php, ESL, and Flex... >> Its not complete by anymeans and you need Flex3 to compile the UI... >> >> Anyone wanting to throw some patches at it for other functionality are >> welcome to do so... One thing is severly lacks at this time is ANY sort of >> authentication...so you wouldn?t want it publically open to the world >> >> K >> >> >> ________________________________ >> From: Harry FSwitch >> Reply-To: >> Date: Mon, 9 Mar 2009 18:17:41 -0400 >> To: >> Subject: [Freeswitch-users] RPC and web admin panel for conference? >> >> Hi all, >> >> I'm looking to implement an admin panel much like the one used at >> http://conference.freeswitch.org. Now I obviously cannot login and see the >> "admin" part of the panel but I'm pretty sure whats in there. >> >> I have xml_rpc running and can connect via http and issue commands. I've >> searched the forum here and went through the wiki, found nothing that looked >> like a panel. I was hoping to find a panel I can just configure and >> implement. Does anyone have a php (I guess, seeing as I have a php server) >> panel they can share with me? I'm sure I can get it working for my system. >> The thought of attempting one on my own at THIS point seems daunting at >> best. >> >> Any help would be greatly appreciated! >> >> Thanks >> >> ________________________________ >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From itche at bridgeport.edu Thu Mar 12 12:26:49 2009 From: itche at bridgeport.edu (itche at bridgeport.edu) Date: Thu, 12 Mar 2009 15:26:49 -0400 (EDT) Subject: [Freeswitch-users] javascript debug Message-ID: <3306.69.120.166.49.1236886009.squirrel@webmail.bridgeport.edu> Hello all, I'm new to this very nice system I'm looking into writing javascripts to interact with the system. How can one debug, run step by step and get variables values the javascripts running under this system? For example I have setup the sample script that do the "tone tests" but I cant find a way to single step it and to look into it. Thanks Itche From msc at freeswitch.org Thu Mar 12 12:59:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Mar 2009 12:59:13 -0700 Subject: [Freeswitch-users] javascript debug In-Reply-To: <3306.69.120.166.49.1236886009.squirrel@webmail.bridgeport.edu> References: <3306.69.120.166.49.1236886009.squirrel@webmail.bridgeport.edu> Message-ID: <87f2f3b90903121259j767ed323la52035b93e9f20d7@mail.gmail.com> On Thu, Mar 12, 2009 at 12:26 PM, wrote: > Hello all, > I'm new to this very nice system > I'm looking into writing javascripts to interact with the system. > How can one debug, run step by step and get variables values the > javascripts running under this system? For example I have setup the sample > script that do the "tone tests" but I cant find a way to single step it > and to look into it. I think you're unable to step through scripts that are launched "inside" of FS. Your best bet is to use lots of console log messages. I prefer to use the INFO level so that I don't see all the debug messages flying by: console_log("info", "This line should appear in green letters on the FS CLI.\n"); You can also print the values of variables using this function. -MC From sprice at gmail.com Thu Mar 12 13:07:44 2009 From: sprice at gmail.com (SP) Date: Thu, 12 Mar 2009 15:07:44 -0500 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! In-Reply-To: References: Message-ID: <7e2ac3270903121307x790cc236k56792f5c655702e8@mail.gmail.com> Another jab, in fun: hey look, that first link looks like a forum?? wow :D On Thu, Mar 12, 2009 at 13:55, Harry FSwitch wrote: > Greetings, > > Last week I submitted this post to the mailing list... > > http://www.nabble.com/Please-end-the-torment-td22352222.html > -- Shannon From mszlazak at aol.com Thu Mar 12 14:22:28 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 12 Mar 2009 17:22:28 -0400 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! In-Reply-To: References: Message-ID: <8CB716FC809DF3B-8FC-1005@webmail-mh34.sysops.aol.com> Great, I like forums better than lists. Make sure the folks at FreeSwitch make it know to anyone coming to any of their pages by providing links, etc. Thanks -----Original Message----- From: Harry FSwitch To: freeswitch-users at lists.freeswitch.org Sent: Thu, 12 Mar 2009 11:55 am Subject: [Freeswitch-users] FreeSWITCH forum community opened today! Greetings, Last week I submitted this post to the mailing list... http://www.nabble.com/Please-end-the-torment-td22352222.html I received a mixed response to say the least, no one however emailed me saying they wanted to be involved or would contribute hosting resources. I figured eh screw-it, and went ahead and created the forums anyway and opened them today! http://freeswitch411.info I went on IRC to let folks know its available and after a few jabs from the crowd Anthony said: "#freeswitch 2009-03-12 10:47:04 [anthm] harr, if you maintain it you are welcome to have it" If it takes off and provides a friendly, helpful entryway for new FreeSWITCH users then I will be happy, if it flops I'll be said. But I will maintain it and do what I can to help FreeSWITCH grow. This is the only time I'll overtly mention it on this list, I'll have the link in my signature and thats about it. :) Thanks for your attention -- Harry http://freeswitch411.info _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/1c17c320/attachment-0002.html From qulix at mail.ru Thu Mar 12 14:26:52 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Fri, 13 Mar 2009 00:26:52 +0300 Subject: [Freeswitch-users] Freeswitch wav-files playback. Message-ID: Greetings People! I am starting user of freeswitch and that is my first time I met such problem. My installed pack (FreeSWITCH Version 1.0.trunk (12573).) doesnt play any wav files. That is what I recieve, when Im trying to playback 2009-03-13 00:14:26 [ERR] switch_core_file.c:71 switch_core_perform_file_open() Invalid file format [wav] for [/usr/local/freeswitch/sounds/en/us/callie/1.wav]! Ofcourse the file exists, and I played it on another work freeswitch station - and it sound. _____ at _______:/usr/local/freeswitch/sounds/en/us/callie# file 1.wav 1.wav: RIFF (little-endian) data, WAVE audio, IMA ADPCM, mono 16000 Hz My OS is Ubuntu 8.04 server. PS. One thing - no metter does wav file exist or not - the error always the same. =\ Tryied more files - same result. I also tryied to reconfigure/remake it - doesnt help. Maybe getting new trunk will help? Thanks. From brian at freeswitch.org Thu Mar 12 14:43:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Mar 2009 16:43:35 -0500 Subject: [Freeswitch-users] Freeswitch wav-files playback. In-Reply-To: References: Message-ID: <522EF23F-D3C1-433B-B887-8105851A1234@freeswitch.org> Sounds like you don' t have mod_sndfile loaded. /b On Mar 12, 2009, at 4:26 PM, ????... wrote: > _____ at _______:/usr/local/freeswitch/sounds/en/us/callie# file 1.wav > 1.wav: RIFF (little-endian) data, WAVE audio, IMA ADPCM, mono 16000 Hz From wiltingtree at gmail.com Thu Mar 12 15:39:27 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Thu, 12 Mar 2009 18:39:27 -0400 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails Message-ID: Sorry, yes I mean cellular phones. Is there some way to notify them of new voicemail messages? > > Message: 7 > Date: Thu, 12 Mar 2009 00:09:35 -0500 > From: Ken Rice > Subject: Re: [Freeswitch-users] How to notify wireless phones about > unread voicemails > To: > Message-ID: > > > Content-Type: text/plain; charset="us-ascii" > > When you say wireless do you mean like Cellular Phone? > > > > From: Adam Wilt > Reply-To: > Date: Thu, 12 Mar 2009 01:02:46 -0400 > To: > Subject: [Freeswitch-users] How to notify wireless phones about unread > voicemails > > Hi, I'm trying to use FreeSWITCH's mod_voicemail to replace the voicemail > on > wireless phones. Does anybody know how to make the wireless phone know > there > is a voicemail waiting, so it can notify the user? > Thanks for the help! > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/4b5624bc/attachment-0002.html From rupa at rupa.com Thu Mar 12 15:45:00 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 12 Mar 2009 17:45:00 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: References: Message-ID: Send them an SMS? On Thu, Mar 12, 2009 at 5:39 PM, Adam Wilt wrote: > Sorry, yes I mean cellular phones. Is there some way to notify them of new > voicemail messages? > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/50431e20/attachment-0002.html From brian at freeswitch.org Thu Mar 12 15:48:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Mar 2009 17:48:52 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: References: Message-ID: <22A9E1FD-232E-4152-9441-EF8E98D04B22@freeswitch.org> Yes there is a way and If I could recall exactly how.. its binary data in the SMS... but some providers IE AT&T block binary sms... so its pointless.. works with T-Mobile... You can toggle the vm, email, fax and other flags on and off via SMS. /b On Mar 12, 2009, at 5:45 PM, Rupa Schomaker wrote: > Send them an SMS? > > On Thu, Mar 12, 2009 at 5:39 PM, Adam Wilt > wrote: > Sorry, yes I mean cellular phones. Is there some way to notify them > of new voicemail messages? > > > > -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/48476cee/attachment-0002.html From krice at freeswitch.org Thu Mar 12 16:10:26 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 12 Mar 2009 18:10:26 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unread voicemails In-Reply-To: Message-ID: Theres the SMS method that other people have mention... There are other methods if you are the Cellular Provider for say a CoOp or a rural ilec... These include (but not limited too and depend on your switch) SS7 TCAP MWI message, sending the correct DTMF down the loop to the switch, etc etc... That being said its posible to use freeswitch for this... It just depends on exactly what type of interconnect you are doing From: Adam Wilt Reply-To: Date: Thu, 12 Mar 2009 18:39:27 -0400 To: Subject: Re: [Freeswitch-users] How to notify wireless phones about unread voicemails Sorry, yes I mean cellular phones. Is there some way to notify them of new voicemail messages? ? > > Message: 7 > Date: Thu, 12 Mar 2009 00:09:35 -0500 > From: Ken Rice > Subject: Re: [Freeswitch-users] How to notify wireless phones about > ? ? ? ?unread voicemails > To: > Message-ID: > > Content-Type: text/plain; charset="us-ascii" > > When you say wireless do you mean like Cellular Phone? > > > > From: Adam Wilt > Reply-To: > Date: Thu, 12 Mar 2009 01:02:46 -0400 > To: > Subject: [Freeswitch-users] How to notify wireless phones about unread > voicemails > > Hi, I'm trying to use FreeSWITCH's mod_voicemail to replace the voicemail on > wireless phones. Does anybody know how to make the wireless phone know there > is a voicemail waiting, so it can notify the user? > Thanks for the help! > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/e9a69882/attachment-0002.html From yudha2008 at gmail.com Thu Mar 12 21:58:34 2009 From: yudha2008 at gmail.com (Baskar) Date: Fri, 13 Mar 2009 10:28:34 +0530 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> References: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> Message-ID: *Hi Michael Collins, When i load mod_openzap i can able to get this output **Successfully Loaded [mod_openzap**] with one error message 2009-03-13 10:21:42 [ERR] mod_openzap.c:1898 load_config() Error starting OpenZAP span 1 mode: -1264601207 dialect: -1264601162 error: I have pasted full output in this path : http://pastebin.freeswitch.org/7746 For oz list command i get this output.* * oz list API CALL [oz(list)] output: For oz dump 1 command i get this output: oz dump 1 API CALL [oz(dump 1)] output: -ERR invalid span Correct where i am worng. ** -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/c4603bc4/attachment-0002.html From asannucci at gmail.com Thu Mar 12 22:33:45 2009 From: asannucci at gmail.com (Andrea) Date: Fri, 13 Mar 2009 00:33:45 -0500 Subject: [Freeswitch-users] How to notify wireless phones about unreadvoicemails References: <22A9E1FD-232E-4152-9441-EF8E98D04B22@freeswitch.org> Message-ID: <9FF857FB3FC345918FD05DD8B756B1EA@quos> With some nokia models (with SIP client) the voicemail notify work fine. I tried it with FS with a Nokia 6300i - Andrea - -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/187d13c5/attachment-0002.html From mszlazak at aol.com Thu Mar 12 23:38:53 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Fri, 13 Mar 2009 02:38:53 -0400 Subject: [Freeswitch-users] How do I notify FreeSwitch that a phone has been answered to play audio or TTS Message-ID: <8CB71BD82A0E313-CF4-158F@MBLK-M31.sysops.aol.com> If I originate an outgoing call from FreeSwitch and want to tts a phrase or play an audio once the call has been answered (i.e, someone answered their cell phone or I got their voicemail) then how do I detect that. Otherwise, I've tried the following but it relies in getting the timing right which won't always work or looping the tts phrase over and over. var s; while (tryCalling()) {} s.hangup(); exit(); function tryCalling() { ??? s = new Session("sofia/gateway/spa3102/12223334444 at 10.0.0.5:5061"); ??? s.waitForAnswer(10000); ??? ??? if (s.cause == "USER_BUSY") { ??? ?? ?return true; ??? } ??? if (s.ready()) { ??? ?? ?s.sleep(10000); ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); ??? } ??? return false; } Another way is to keep replaying the tts phrase by replacing ? if (s.ready()) { ??? ?? ?s.sleep(10000); ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); ??? } with something like: while (s.ready()) { ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); ??? } Thanks. Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/4574725b/attachment-0002.html From jalsot at gmail.com Fri Mar 13 01:46:39 2009 From: jalsot at gmail.com (Tamas) Date: Fri, 13 Mar 2009 09:46:39 +0100 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: References: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> Message-ID: <49BA1D6F.30002@gmail.com> Hello, please show your configs (openzap.conf, openzap.conf.xml). Which protocol do you intend to use? (EuroISDN, etc.) Regards, Tamas Baskar ?rta: > *Hi Michael Collins, > > When i load mod_openzap i can able to get this output > **Successfully Loaded [mod_openzap**] with one error message > > 2009-03-13 10:21:42 [ERR] mod_openzap.c:1898 load_config() Error > starting OpenZAP span 1 mode: -1264601207 dialect: -1264601162 error: > > I have pasted full output in this path : > http://pastebin.freeswitch.org/7746 > > For oz list command i get this output.* > > * oz list > API CALL [oz(list)] output: > > For oz dump 1 command i get this output: > > oz dump 1 > API CALL [oz(dump 1)] output: > -ERR invalid span > > Correct where i am worng. > ** > -- > Warm Regards, > N.Baskar > * > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kawarod at laposte.net Fri Mar 13 01:51:49 2009 From: kawarod at laposte.net (rod) Date: Fri, 13 Mar 2009 12:51:49 +0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49B93321.5080500@gmail.com> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> Message-ID: <49BA1EA5.4050201@laposte.net> Hello, I'm now running r12590, the pbm was still there, but this was because of a broken dialplan. I'm using this for exceeded limit: but this extension was at the end of my dialplan and I matched an other extension before reaching the limit extension. What was odd was that in ngrep I saw FS sending a 503, I will investigate on this. I will rerun long term test and let you know if all is ok, so I have to do for a previous mail related to ghost sessions in CLI. @Tamas: I just give a try to limit_hash, and didn't make many tests with it. It's on my todo list. Limit_hash is not a better choice than limit, their usage are different: - limit_hash is a good way to rate limit a specific gateway for example so that your switch won't be flooded by a misconfigured peer gw - limit is for limiting concurrent call to an extension/gateway, eg you have a peer that provides you with 30channels and you want to allow 15 channels for mobile (PLMN) and 15 for PSTN without relying If you still have pbm with limit and svn, pay attention to you dialplan :p @Mathieu I hope you didn't work on a virtual pbm, cause it seems to be a dialplan misconfiguration. I'll let you know if I still have pbm. Thanks for your help. regards. rod Tamas Cseke wrote: > Hello, > > we had the same problem. we couldn't test r12581 for sure yet, but we will. > This fix is only for the limit (db version), right? > Would be limit_hash a better choice to increase performance anyway? > Rod, do yoo have maybe experiences with limit_hash with your sipp? > > Thanks in advance, > Tamas > > > Mathieu Rene ?rta: > >> Were you doing transfers (action="transfer" in the dialplan) ? >> >> If yes, retry with revision 12581, and open a JIRA if it still is an >> issue (also include a full debug log (delete your freeswitch.log file >> before doing your test and attach it after) in your JIRA) >> >> Math >> >> On 12-Mar-09, at 10:12 AM, rod wrote: >> >> >> >>> Hi list, >>> >>> I'm testing mod_limit like this: >>> >>> >>> >>> >>> where AREA could be any value like: US/EUROPE/AFRICA... that has been >>> set for the call >>> >>> then I use sipp for load testing with 4 cps and max 65 calls so that I >>> will be limited by the limit of 60 in the dialplan. >>> >>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>> value >>> growing up to 60. >>> >>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>> for >>> each new cps in excess, FS sends a 503. >>> I wait for 10 seconds and stop sipp, but the limit value is never >>> decreasing even when there is no more channels used (show channels), >>> the >>> limit value is stuck to 60. >>> >>> >>> If I limit Sipp to 55 calls (below the limit value), the limit value >>> increase and decrease depending on load, and the pbm doesn't appear. >>> >>> Does anybody is using mod_limit and have encountered the same pbm. >>> >>> I'm using latest svn: 12580. >>> >>> regards, >>> rod >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From Richard.Lamkin at mettoni.com Fri Mar 13 03:09:58 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Fri, 13 Mar 2009 10:09:58 -0000 Subject: [Freeswitch-users] Sending SUBCRIBE request to a peer PBX Message-ID: <3181A30B8C35AB4AA8577B78DDF461380499DBC8@nickel.mettonigroup.com> Dear All, I have an external gateway defined which registers as a client to another SIP PBX [a Nortel CS2K]. In effect FS is an extension of the CS2K. The CS2K can provide presence info of the other extensions using the SUBSCRIBE method. Is there a way to issue SUBSCIBE and handle [record or forward as an event] the returned NOTIFY. An example would be using Xlite as the client and it can harvest the presence info for softphones in the contact list. I have tested that FS delivers presence info to clients of it. Regards Richard Lamkin ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/529b97ca/attachment-0002.html From regs at kinetix.gr Fri Mar 13 06:25:08 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Fri, 13 Mar 2009 15:25:08 +0200 Subject: [Freeswitch-users] Call-Direction issue Message-ID: <49BA5EB4.7040907@kinetix.gr> Hi, I am trying to include the Call-Direction variable (${direction}) in my csv cdrs. But I get nothing. I tried printing the variable in the console before a bridge : But nothing there too. When I use uuid_dump during a call I can see the Call-Direction variable. I tried call_direction as well but it doesn't work either. Any ideas? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From Richard.Lamkin at mettoni.com Fri Mar 13 06:29:25 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Fri, 13 Mar 2009 13:29:25 -0000 Subject: [Freeswitch-users] Load testing on XP Message-ID: <3181A30B8C35AB4AA8577B78DDF461380499DD68@nickel.mettonigroup.com> Dear All, Does anyone have any load test results using XP(SP3) - I have checked the wiki but could not find any recorded. Regards Richard Lamkin ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/f2911933/attachment-0002.html From mike at jerris.com Fri Mar 13 06:33:20 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 13 Mar 2009 09:33:20 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA1EA5.4050201@laposte.net> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> Message-ID: <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> What is a pbm? On Mar 13, 2009, at 4:51 AM, rod wrote: > Hello, > > I'm now running r12590, the pbm was still there, but this was > because of > a broken dialplan. > > I'm using this for exceeded limit: > > expression="^limit_exceeded$"> > > > > > > but this extension was at the end of my dialplan and I matched an > other > extension before reaching the limit extension. > What was odd was that in ngrep I saw FS sending a 503, I will > investigate on this. > > I will rerun long term test and let you know if all is ok, so I have > to > do for a previous mail related to ghost sessions in CLI. > > @Tamas: > I just give a try to limit_hash, and didn't make many tests with it. > It's on my todo list. Limit_hash is not a better choice than limit, > their usage are different: > - limit_hash is a good way to rate limit a specific gateway for > example so that your switch won't be flooded by a misconfigured peer > gw > - limit is for limiting concurrent call to an extension/gateway, eg > you have a peer that provides you with 30channels and you want to > allow > 15 channels for mobile (PLMN) and 15 for PSTN without relying > > If you still have pbm with limit and svn, pay attention to you > dialplan :p > > @Mathieu > I hope you didn't work on a virtual pbm, cause it seems to be a > dialplan > misconfiguration. I'll let you know if I still have pbm. Thanks for > your > help. > > regards. > rod > > Tamas Cseke wrote: >> Hello, >> >> we had the same problem. we couldn't test r12581 for sure yet, but >> we will. >> This fix is only for the limit (db version), right? >> Would be limit_hash a better choice to increase performance anyway? >> Rod, do yoo have maybe experiences with limit_hash with your sipp? >> >> Thanks in advance, >> Tamas >> >> >> Mathieu Rene ?rta: >> >>> Were you doing transfers (action="transfer" in the dialplan) ? >>> >>> If yes, retry with revision 12581, and open a JIRA if it still is an >>> issue (also include a full debug log (delete your freeswitch.log >>> file >>> before doing your test and attach it after) in your JIRA) >>> >>> Math >>> >>> On 12-Mar-09, at 10:12 AM, rod wrote: >>> >>> >>> >>>> Hi list, >>>> >>>> I'm testing mod_limit like this: >>>> >>>> >>>> >>>> >>>> where AREA could be any value like: US/EUROPE/AFRICA... that has >>>> been >>>> set for the call >>>> >>>> then I use sipp for load testing with 4 cps and max 65 calls so >>>> that I >>>> will be limited by the limit of 60 in the dialplan. >>>> >>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>> value >>>> growing up to 60. >>>> >>>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>>> for >>>> each new cps in excess, FS sends a 503. >>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>> decreasing even when there is no more channels used (show >>>> channels), >>>> the >>>> limit value is stuck to 60. >>>> >>>> >>>> If I limit Sipp to 55 calls (below the limit value), the limit >>>> value >>>> increase and decrease depending on load, and the pbm doesn't >>>> appear. >>>> >>>> Does anybody is using mod_limit and have encountered the same pbm. >>>> >>>> I'm using latest svn: 12580. >>>> >>>> regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From davidwdan at gmail.com Fri Mar 13 06:45:13 2009 From: davidwdan at gmail.com (David Dan) Date: Fri, 13 Mar 2009 09:45:13 -0400 Subject: [Freeswitch-users] How do I notify FreeSwitch that a phone has been answered to play audio or TTS In-Reply-To: <8CB71BD82A0E313-CF4-158F@MBLK-M31.sysops.aol.com> References: <8CB71BD82A0E313-CF4-158F@MBLK-M31.sysops.aol.com> Message-ID: <65bd1c9f0903130645t70c12458n563319f85fe0cf3a@mail.gmail.com> Try ignore early media. On 3/13/09, mszlazak at aol.com wrote: > If I originate an outgoing call from FreeSwitch and want to tts a phrase or > play an audio once the call has been answered (i.e, someone answered their > cell phone or I got their voicemail) then how do I detect that. > > Otherwise, I've tried the following but it relies in getting the timing > right which won't always work or looping the tts phrase over and over. > > var s; > while (tryCalling()) {} > s.hangup(); > exit(); > > function tryCalling() { > ??? s = new Session("sofia/gateway/spa3102/12223334444 at 10.0.0.5:5061"); > ??? s.waitForAnswer(10000); > ??? > ??? if (s.cause == "USER_BUSY") { > ??? ?? ?return true; > ??? } > > ??? if (s.ready()) { > ??? ?? ?s.sleep(10000); > ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); > ??? } > ??? return false; > } > > > Another way is to keep replaying the tts phrase by replacing > > ? if (s.ready()) { > > ??? ?? ?s.sleep(10000); > > ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); > > ??? } > > with something like: > > while (s.ready()) { > > ??? ?? ?s.speak("cepstral","Callie","Hello from FreeSwitch"); > > ??? } > > > Thanks. Mark. > -- Sent from my mobile device From jalsot at gmail.com Fri Mar 13 06:56:24 2009 From: jalsot at gmail.com (Tamas) Date: Fri, 13 Mar 2009 14:56:24 +0100 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> Message-ID: <49BA6608.70208@gmail.com> My guess is: pbm = problem :) T. Michael Jerris ?rta: > What is a pbm? > > On Mar 13, 2009, at 4:51 AM, rod wrote: > > >> Hello, >> >> I'm now running r12590, the pbm was still there, but this was >> because of >> a broken dialplan. >> >> I'm using this for exceeded limit: >> >> > expression="^limit_exceeded$"> >> >> >> >> >> >> but this extension was at the end of my dialplan and I matched an >> other >> extension before reaching the limit extension. >> What was odd was that in ngrep I saw FS sending a 503, I will >> investigate on this. >> >> I will rerun long term test and let you know if all is ok, so I have >> to >> do for a previous mail related to ghost sessions in CLI. >> >> @Tamas: >> I just give a try to limit_hash, and didn't make many tests with it. >> It's on my todo list. Limit_hash is not a better choice than limit, >> their usage are different: >> - limit_hash is a good way to rate limit a specific gateway for >> example so that your switch won't be flooded by a misconfigured peer >> gw >> - limit is for limiting concurrent call to an extension/gateway, eg >> you have a peer that provides you with 30channels and you want to >> allow >> 15 channels for mobile (PLMN) and 15 for PSTN without relying >> >> If you still have pbm with limit and svn, pay attention to you >> dialplan :p >> >> @Mathieu >> I hope you didn't work on a virtual pbm, cause it seems to be a >> dialplan >> misconfiguration. I'll let you know if I still have pbm. Thanks for >> your >> help. >> >> regards. >> rod >> >> Tamas Cseke wrote: >> >>> Hello, >>> >>> we had the same problem. we couldn't test r12581 for sure yet, but >>> we will. >>> This fix is only for the limit (db version), right? >>> Would be limit_hash a better choice to increase performance anyway? >>> Rod, do yoo have maybe experiences with limit_hash with your sipp? >>> >>> Thanks in advance, >>> Tamas >>> >>> >>> Mathieu Rene ?rta: >>> >>> >>>> Were you doing transfers (action="transfer" in the dialplan) ? >>>> >>>> If yes, retry with revision 12581, and open a JIRA if it still is an >>>> issue (also include a full debug log (delete your freeswitch.log >>>> file >>>> before doing your test and attach it after) in your JIRA) >>>> >>>> Math >>>> >>>> On 12-Mar-09, at 10:12 AM, rod wrote: >>>> >>>> >>>> >>>> >>>>> Hi list, >>>>> >>>>> I'm testing mod_limit like this: >>>>> >>>>> >>>>> >>>>> >>>>> where AREA could be any value like: US/EUROPE/AFRICA... that has >>>>> been >>>>> set for the call >>>>> >>>>> then I use sipp for load testing with 4 cps and max 65 calls so >>>>> that I >>>>> will be limited by the limit of 60 in the dialplan. >>>>> >>>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>>> value >>>>> growing up to 60. >>>>> >>>>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>>>> for >>>>> each new cps in excess, FS sends a 503. >>>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>>> decreasing even when there is no more channels used (show >>>>> channels), >>>>> the >>>>> limit value is stuck to 60. >>>>> >>>>> >>>>> If I limit Sipp to 55 calls (below the limit value), the limit >>>>> value >>>>> increase and decrease depending on load, and the pbm doesn't >>>>> appear. >>>>> >>>>> Does anybody is using mod_limit and have encountered the same pbm. >>>>> >>>>> I'm using latest svn: 12580. >>>>> >>>>> regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Gabriel at airg.com Thu Mar 12 15:28:02 2009 From: Gabriel at airg.com (Gabriel Cho) Date: Thu, 12 Mar 2009 15:28:02 -0700 Subject: [Freeswitch-users] uuid_displace in bridged call Message-ID: <0B02E756F603CC409EB553879B090CC80A50E0C6@HPEXCHVS01.exchange.airg> Hey all, does anyone know how to detect a completion of an audio file playback using uuid_displace API in a bridged call? ie) socket.send("api uuid_bridge #{uuid} #{other_uuid}"); socket.send("api uuid_displace #{uuid} start #{filename} 0 mux"); <-- i do not receive any events from freeswitch notifying end of audio file Gabriel Cho -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090312/93acb4a5/attachment-0002.html From diego.viola at gmail.com Fri Mar 13 03:46:09 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 06:46:09 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime Message-ID: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> Hello, I was wondering if it is possible to start FreeSWITCH without any SQL at all, yeah I know there is -nosql but I still want to have all the information available at run time, ie: show channels, sofia status, sofia status profile , everything. Would that improve performance? I ask this because I heard that SQLite causes some performance issues and I heard it's being removed from the core, I also think that making that optional would be nice along with UnixODBC support for everything. I also got a few situations where I had to rm -f /usr/local/freeswitch/db/* because there were some data there that was not supposed to be, that's why I think real time data on run time would be better. Anyway, just some feedback/question I wanted to share, FS is awesome, keep up the great work. Regards, Diego From diego.viola at gmail.com Fri Mar 13 03:53:01 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 06:53:01 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> Message-ID: <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> This is what I get when I do a show channels with -nosql. freeswitch at internal> show channels -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! freeswitch at internal> It would be nice if it prints the channels on runtime without the need/overheat of SQL. Regards, Diego On Fri, Mar 13, 2009 at 6:46 AM, Diego Viola wrote: > Hello, > > I was wondering if it is possible to start FreeSWITCH without any SQL > at all, yeah I know there is -nosql but I still want to have all the > information available at run time, ie: show channels, sofia status, > sofia status profile , everything. > > Would that improve performance? I ask this because I heard that SQLite > causes some performance issues and I heard it's being removed from the > core, I also think that making that optional would be nice along with > UnixODBC support for everything. > > I also got a few situations where I had to rm -f > /usr/local/freeswitch/db/* because there were some data there that was > not supposed to be, that's why I think real time data on run time > would be better. > > Anyway, just some feedback/question I wanted to share, FS is awesome, > keep up the great work. > > Regards, > > Diego > From diego.viola at gmail.com Fri Mar 13 04:12:17 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 07:12:17 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> Message-ID: <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> Erhm, this is what I wanted to say. -- I think FreeSWITCH should leave that SQLite option as optional, I think it would be nicer if FS doesn't depend on any SQL at all, if you do a show channels right now with -nosql, this is what you will see: freeswitch at internal> show channels -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! freeswitch at internal> I think it would be nice if FS can show you the information on real time, and on run time, without any SQL at all, and leave UnixODBC & SQLite as optional options, I also think this will improve performance. On Fri, Mar 13, 2009 at 6:53 AM, Diego Viola wrote: > This is what I get when I do a show channels with -nosql. > > freeswitch at internal> show channels > -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! > > freeswitch at internal> > > It would be nice if it prints the channels on runtime without the > need/overheat of SQL. > > Regards, > > Diego > > On Fri, Mar 13, 2009 at 6:46 AM, Diego Viola wrote: >> Hello, >> >> I was wondering if it is possible to start FreeSWITCH without any SQL >> at all, yeah I know there is -nosql but I still want to have all the >> information available at run time, ie: show channels, sofia status, >> sofia status profile , everything. >> >> Would that improve performance? I ask this because I heard that SQLite >> causes some performance issues and I heard it's being removed from the >> core, I also think that making that optional would be nice along with >> UnixODBC support for everything. >> >> I also got a few situations where I had to rm -f >> /usr/local/freeswitch/db/* because there were some data there that was >> not supposed to be, that's why I think real time data on run time >> would be better. >> >> Anyway, just some feedback/question I wanted to share, FS is awesome, >> keep up the great work. >> >> Regards, >> >> Diego >> > From diego.viola at gmail.com Fri Mar 13 05:56:46 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 08:56:46 -0400 Subject: [Freeswitch-users] FreeSWITCH forum community opened today! In-Reply-To: <8CB716FC809DF3B-8FC-1005@webmail-mh34.sysops.aol.com> References: <8CB716FC809DF3B-8FC-1005@webmail-mh34.sysops.aol.com> Message-ID: <86a32abc0903130556r5ea96a6anb47eb99b7052756b@mail.gmail.com> Good initiative, congrats. I already registered to it. Diego On Thu, Mar 12, 2009 at 5:22 PM, wrote: > Great, I like forums better than lists. Make sure the folks at FreeSwitch > make it know to anyone coming to any of their pages by providing links, etc. > Thanks > > > -----Original Message----- > From: Harry FSwitch > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, 12 Mar 2009 11:55 am > Subject: [Freeswitch-users] FreeSWITCH forum community opened today! > > Greetings, > > Last week I submitted this post to the mailing list... > > http://www.nabble.com/Please-end-the-torment-td22352222.html > > I received a mixed response to say the least, no one however emailed me > saying they wanted to be involved or would contribute hosting resources. I > figured eh screw-it, and went ahead and created the forums anyway and opened > them today! > > http://freeswitch411.info > > I went on IRC to let folks know its available and after a few jabs from the > crowd Anthony said: > "#freeswitch 2009-03-12 10:47:04 [anthm] harr, if you maintain it you are > welcome to have it" > > If it takes off and provides a friendly, helpful entryway for new FreeSWITCH > users then I will be happy, if it flops I'll be said. But I will maintain it > and do what I can to help FreeSWITCH grow. This is the only time I'll > overtly mention it on this list, I'll have the link in my signature and > thats about it. :) > > Thanks for your attention > > -- > Harry > http://freeswitch411.info > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ________________________________ > A Good Credit Score is 700 or Above. See yours in just 2 easy steps! > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From rupa at rupa.com Fri Mar 13 07:05:33 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 13 Mar 2009 09:05:33 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> Message-ID: Using sql for the show commands is pretty elegant. It allows one to get info without having to lock a bunch of in memory data structures prior to collecting stats. Locking data structures just for reporting == bad -- especially on a high volume server. On Fri, Mar 13, 2009 at 6:12 AM, Diego Viola wrote: > Erhm, this is what I wanted to say. > > -- > > I think FreeSWITCH should leave that SQLite option as optional, I > think it would be nicer if FS doesn't depend on any SQL at all, if you > do a show channels right now with -nosql, this is what you will see: > > freeswitch at internal> show channels > -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! > > freeswitch at internal> > > I think it would be nice if FS can show you the information on real > time, and on run time, without any SQL at all, and leave UnixODBC & > SQLite as optional options, I also think this will improve > performance. > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/bf0a7d26/attachment-0002.html From mrene_lists at avgs.ca Fri Mar 13 07:09:05 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 13 Mar 2009 10:09:05 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> Message-ID: <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> In this case sqlite separates the core from what show channels uses etc, the DB is filled in by an event handler. This actually increases performance since you dont need to exclusively lock the session manager's table just to list channels. Math On 13-Mar-09, at 7:12 AM, Diego Viola wrote: > Erhm, this is what I wanted to say. > > -- > > I think FreeSWITCH should leave that SQLite option as optional, I > think it would be nicer if FS doesn't depend on any SQL at all, if you > do a show channels right now with -nosql, this is what you will see: > > freeswitch at internal> show channels > -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! > > freeswitch at internal> > > I think it would be nice if FS can show you the information on real > time, and on run time, without any SQL at all, and leave UnixODBC & > SQLite as optional options, I also think this will improve > performance. > > On Fri, Mar 13, 2009 at 6:53 AM, Diego Viola > wrote: >> This is what I get when I do a show channels with -nosql. >> >> freeswitch at internal> show channels >> -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! >> >> freeswitch at internal> >> >> It would be nice if it prints the channels on runtime without the >> need/overheat of SQL. >> >> Regards, >> >> Diego >> >> On Fri, Mar 13, 2009 at 6:46 AM, Diego Viola >> wrote: >>> Hello, >>> >>> I was wondering if it is possible to start FreeSWITCH without any >>> SQL >>> at all, yeah I know there is -nosql but I still want to have all the >>> information available at run time, ie: show channels, sofia status, >>> sofia status profile , everything. >>> >>> Would that improve performance? I ask this because I heard that >>> SQLite >>> causes some performance issues and I heard it's being removed from >>> the >>> core, I also think that making that optional would be nice along >>> with >>> UnixODBC support for everything. >>> >>> I also got a few situations where I had to rm -f >>> /usr/local/freeswitch/db/* because there were some data there that >>> was >>> not supposed to be, that's why I think real time data on run time >>> would be better. >>> >>> Anyway, just some feedback/question I wanted to share, FS is >>> awesome, >>> keep up the great work. >>> >>> Regards, >>> >>> Diego >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Mark.Tabron at rnid-typetalk.org.uk Fri Mar 13 07:16:14 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Fri, 13 Mar 2009 14:16:14 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> Message-ID: <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! http://pastebin.freeswitch.org/7751 Will setup an IRC client and see if I can log onto the channel. Thanks again! -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 March 2009 16:50 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 > My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so > apologies if some of my terminology isn't quite correct. Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > > > > Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've > hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the > card is green and wanrouter is installed and up in TDM_API mode, with the > connection status showing as connected. ?Configured Openzap for 9 b and 1 d > channel as described in Freeswitch Wiki. Then created a diaplan to fire off > any calls preceded by 9 to the next available openzap channel. Looks good so far... > The problem I have is when I initiate an external call (using 9xxxxxxx) from > an extension I can see Freeswitch allocating the call to the next available > channel but then the just sits there and times out after 1 minute. With the > cause stated as ORIGINATOR_CANCEL (guessing this is the time out) okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has lots of useful information for how to gather log information, how to use the pastebin, etc. Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers: To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org). Welcome to the wonderful world of telephony debugging! -MC P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.) _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- From intralanman at freeswitch.org Fri Mar 13 09:05:32 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 13 Mar 2009 12:05:32 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA6608.70208@gmail.com> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> <49BA6608.70208@gmail.com> Message-ID: <49BA844C.3010409@freeswitch.org> Tamas wrote: > My guess is: pbm = problem :) > sure, but is it really that hard to spell all the way out? -Ray From intralanman at freeswitch.org Fri Mar 13 09:09:39 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 13 Mar 2009 12:09:39 -0400 Subject: [Freeswitch-users] Load testing on XP In-Reply-To: <3181A30B8C35AB4AA8577B78DDF461380499DD68@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF461380499DD68@nickel.mettonigroup.com> Message-ID: <49BA8543.5070401@freeswitch.org> Richard Lamkin wrote: > > Dear All, > > > > Does anyone have any load test results using XP(SP3) -- I have checked > the wiki but could not find any recorded. > > > > Regards > > > > Richard Lamkin > for the most part, we try to steer clear of posting load test results since my testing may be different than yours, it's hard (if not impossible) for two separate users to get the same results. so we recommend you try your own load testing the mirrors your scenario specifically. -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/8d11e376/attachment-0002.html From cstomi.levlist at gmail.com Fri Mar 13 09:19:00 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Fri, 13 Mar 2009 17:19:00 +0100 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA1EA5.4050201@laposte.net> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> Message-ID: <49BA8774.3040803@gmail.com> Hello, Thank you for your help. limit,Limit, [ [number [dialplan [context]]]] limit_hash,Limit (hash), [[/interval]] [number [dialplan [context]]] I diged into mod_limit deeplier and as far as I understand, limit and limit_hash can be used for the same thing: limiting concurrent calls. please correct me, if I'm wrong... but limit_hash has another feature /interval this is that make the rate limitation you mentioned. Our dialplan is good that is out of question. I see hangup in console. we do I tested both of them with sipp and I wasn't able to reproduce the problem r12581 did fix the case when limit is called more times, right? Thanks, Tamas rod ?rta: > Hello, > > I'm now running r12590, the pbm was still there, but this was because of > a broken dialplan. > > I'm using this for exceeded limit: > > > > > > > > but this extension was at the end of my dialplan and I matched an other > extension before reaching the limit extension. > What was odd was that in ngrep I saw FS sending a 503, I will > investigate on this. > > I will rerun long term test and let you know if all is ok, so I have to > do for a previous mail related to ghost sessions in CLI. > > @Tamas: > I just give a try to limit_hash, and didn't make many tests with it. > It's on my todo list. Limit_hash is not a better choice than limit, > their usage are different: > - limit_hash is a good way to rate limit a specific gateway for > example so that your switch won't be flooded by a misconfigured peer gw > - limit is for limiting concurrent call to an extension/gateway, eg > you have a peer that provides you with 30channels and you want to allow > 15 channels for mobile (PLMN) and 15 for PSTN without relying > > If you still have pbm with limit and svn, pay attention to you dialplan :p > > @Mathieu > I hope you didn't work on a virtual pbm, cause it seems to be a dialplan > misconfiguration. I'll let you know if I still have pbm. Thanks for your > help. > > regards. > rod > > Tamas Cseke wrote: > >> Hello, >> >> we had the same problem. we couldn't test r12581 for sure yet, but we will. >> This fix is only for the limit (db version), right? >> Would be limit_hash a better choice to increase performance anyway? >> Rod, do yoo have maybe experiences with limit_hash with your sipp? >> >> Thanks in advance, >> Tamas >> >> >> Mathieu Rene ?rta: >> >> >>> Were you doing transfers (action="transfer" in the dialplan) ? >>> >>> If yes, retry with revision 12581, and open a JIRA if it still is an >>> issue (also include a full debug log (delete your freeswitch.log file >>> before doing your test and attach it after) in your JIRA) >>> >>> Math >>> >>> On 12-Mar-09, at 10:12 AM, rod wrote: >>> >>> >>> >>> >>>> Hi list, >>>> >>>> I'm testing mod_limit like this: >>>> >>>> >>>> >>>> >>>> where AREA could be any value like: US/EUROPE/AFRICA... that has been >>>> set for the call >>>> >>>> then I use sipp for load testing with 4 cps and max 65 calls so that I >>>> will be limited by the limit of 60 in the dialplan. >>>> >>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>> value >>>> growing up to 60. >>>> >>>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>>> for >>>> each new cps in excess, FS sends a 503. >>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>> decreasing even when there is no more channels used (show channels), >>>> the >>>> limit value is stuck to 60. >>>> >>>> >>>> If I limit Sipp to 55 calls (below the limit value), the limit value >>>> increase and decrease depending on load, and the pbm doesn't appear. >>>> >>>> Does anybody is using mod_limit and have encountered the same pbm. >>>> >>>> I'm using latest svn: 12580. >>>> >>>> regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mrene_lists at avgs.ca Fri Mar 13 09:26:05 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 13 Mar 2009 12:26:05 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA8774.3040803@gmail.com> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <49BA8774.3040803@gmail.com> Message-ID: Yeah, you should be using limit_hash unless you need data replication on multiple FS boxes since its a lot faster. You can also do On another note, normal_circuit_congestion looks like a more appropriate cause. If you have problems, please open a jira including logs and a detailed description of the required steps to reproduce the problem. Math On 13-Mar-09, at 12:19 PM, Tamas Cseke wrote: > Hello, > > Thank you for your help. > > limit,Limit, [ [number [dialplan [context]]]] > limit_hash,Limit (hash), [[/interval]] [number > [dialplan [context]]] > > I diged into mod_limit deeplier and as far as I understand, limit and > limit_hash can be used for the same thing: > limiting concurrent calls. please correct me, if I'm wrong... > > but limit_hash has another feature /interval this is that make the > rate > limitation you mentioned. > > Our dialplan is good that is out of question. I see hangup in console. > we do > > I tested both of them with sipp and I wasn't able to reproduce the > problem > > r12581 did fix the case when limit is called more times, right? > > Thanks, > Tamas > > > rod ?rta: >> Hello, >> >> I'm now running r12590, the pbm was still there, but this was >> because of >> a broken dialplan. >> >> I'm using this for exceeded limit: >> >> > expression="^limit_exceeded$"> >> >> >> >> >> >> but this extension was at the end of my dialplan and I matched an >> other >> extension before reaching the limit extension. >> What was odd was that in ngrep I saw FS sending a 503, I will >> investigate on this. >> >> I will rerun long term test and let you know if all is ok, so I >> have to >> do for a previous mail related to ghost sessions in CLI. >> >> @Tamas: >> I just give a try to limit_hash, and didn't make many tests with it. >> It's on my todo list. Limit_hash is not a better choice than limit, >> their usage are different: >> - limit_hash is a good way to rate limit a specific gateway for >> example so that your switch won't be flooded by a misconfigured >> peer gw >> - limit is for limiting concurrent call to an extension/gateway, >> eg >> you have a peer that provides you with 30channels and you want to >> allow >> 15 channels for mobile (PLMN) and 15 for PSTN without relying >> >> If you still have pbm with limit and svn, pay attention to you >> dialplan :p >> >> @Mathieu >> I hope you didn't work on a virtual pbm, cause it seems to be a >> dialplan >> misconfiguration. I'll let you know if I still have pbm. Thanks for >> your >> help. >> >> regards. >> rod >> >> Tamas Cseke wrote: >> >>> Hello, >>> >>> we had the same problem. we couldn't test r12581 for sure yet, but >>> we will. >>> This fix is only for the limit (db version), right? >>> Would be limit_hash a better choice to increase performance anyway? >>> Rod, do yoo have maybe experiences with limit_hash with your sipp? >>> >>> Thanks in advance, >>> Tamas >>> >>> >>> Mathieu Rene ?rta: >>> >>> >>>> Were you doing transfers (action="transfer" in the dialplan) ? >>>> >>>> If yes, retry with revision 12581, and open a JIRA if it still is >>>> an >>>> issue (also include a full debug log (delete your freeswitch.log >>>> file >>>> before doing your test and attach it after) in your JIRA) >>>> >>>> Math >>>> >>>> On 12-Mar-09, at 10:12 AM, rod wrote: >>>> >>>> >>>> >>>> >>>>> Hi list, >>>>> >>>>> I'm testing mod_limit like this: >>>>> >>>>> >>>>> >>>>> >>>>> where AREA could be any value like: US/EUROPE/AFRICA... that has >>>>> been >>>>> set for the call >>>>> >>>>> then I use sipp for load testing with 4 cps and max 65 calls so >>>>> that I >>>>> will be limited by the limit of 60 in the dialplan. >>>>> >>>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>>> value >>>>> growing up to 60. >>>>> >>>>> Sipp still tries to establish new call (up to 65 calls) at 4cps >>>>> and >>>>> for >>>>> each new cps in excess, FS sends a 503. >>>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>>> decreasing even when there is no more channels used (show >>>>> channels), >>>>> the >>>>> limit value is stuck to 60. >>>>> >>>>> >>>>> If I limit Sipp to 55 calls (below the limit value), the limit >>>>> value >>>>> increase and decrease depending on load, and the pbm doesn't >>>>> appear. >>>>> >>>>> Does anybody is using mod_limit and have encountered the same pbm. >>>>> >>>>> I'm using latest svn: 12580. >>>>> >>>>> regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cstomi.levlist at gmail.com Fri Mar 13 09:17:46 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Fri, 13 Mar 2009 17:17:46 +0100 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA1EA5.4050201@laposte.net> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> Message-ID: <49BA872A.9030003@gmail.com> Hello, Thank you for your help. limit,Limit, [ [number [dialplan [context]]]] limit_hash,Limit (hash), [[/interval]] [number [dialplan [context]]] I diged into mod_limit deeplier and as far as I understand, limit and limit_hash can be used for the same thing: limiting concurrent calls. please correct me, if I'm wrong... but limit_hash has another feature /interval this is that make the rate limitation you mentioned. Our dialplan is good that is out of question. I see hangup in console. we do I tested both of them with sipp and I wasn't able to reproduce the problem r12581 did fix the case when limit is called more times, right? Thanks, Tamas rod ?rta: > Hello, > > I'm now running r12590, the pbm was still there, but this was because of > a broken dialplan. > > I'm using this for exceeded limit: > > > > > > > > but this extension was at the end of my dialplan and I matched an other > extension before reaching the limit extension. > What was odd was that in ngrep I saw FS sending a 503, I will > investigate on this. > > I will rerun long term test and let you know if all is ok, so I have to > do for a previous mail related to ghost sessions in CLI. > > @Tamas: > I just give a try to limit_hash, and didn't make many tests with it. > It's on my todo list. Limit_hash is not a better choice than limit, > their usage are different: > - limit_hash is a good way to rate limit a specific gateway for > example so that your switch won't be flooded by a misconfigured peer gw > - limit is for limiting concurrent call to an extension/gateway, eg > you have a peer that provides you with 30channels and you want to allow > 15 channels for mobile (PLMN) and 15 for PSTN without relying > > If you still have pbm with limit and svn, pay attention to you dialplan :p > > @Mathieu > I hope you didn't work on a virtual pbm, cause it seems to be a dialplan > misconfiguration. I'll let you know if I still have pbm. Thanks for your > help. > > regards. > rod > > Tamas Cseke wrote: > >> Hello, >> >> we had the same problem. we couldn't test r12581 for sure yet, but we will. >> This fix is only for the limit (db version), right? >> Would be limit_hash a better choice to increase performance anyway? >> Rod, do yoo have maybe experiences with limit_hash with your sipp? >> >> Thanks in advance, >> Tamas >> >> >> Mathieu Rene ?rta: >> >> >>> Were you doing transfers (action="transfer" in the dialplan) ? >>> >>> If yes, retry with revision 12581, and open a JIRA if it still is an >>> issue (also include a full debug log (delete your freeswitch.log file >>> before doing your test and attach it after) in your JIRA) >>> >>> Math >>> >>> On 12-Mar-09, at 10:12 AM, rod wrote: >>> >>> >>> >>> >>>> Hi list, >>>> >>>> I'm testing mod_limit like this: >>>> >>>> >>>> >>>> >>>> where AREA could be any value like: US/EUROPE/AFRICA... that has been >>>> set for the call >>>> >>>> then I use sipp for load testing with 4 cps and max 65 calls so that I >>>> will be limited by the limit of 60 in the dialplan. >>>> >>>> I use "limit_usage EUROPE 10.10.20.100" in cli and I see the limit >>>> value >>>> growing up to 60. >>>> >>>> Sipp still tries to establish new call (up to 65 calls) at 4cps and >>>> for >>>> each new cps in excess, FS sends a 503. >>>> I wait for 10 seconds and stop sipp, but the limit value is never >>>> decreasing even when there is no more channels used (show channels), >>>> the >>>> limit value is stuck to 60. >>>> >>>> >>>> If I limit Sipp to 55 calls (below the limit value), the limit value >>>> increase and decrease depending on load, and the pbm doesn't appear. >>>> >>>> Does anybody is using mod_limit and have encountered the same pbm. >>>> >>>> I'm using latest svn: 12580. >>>> >>>> regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Fri Mar 13 07:36:20 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 13 Mar 2009 10:36:20 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> Message-ID: <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> Oh, I thought that SQLite wasn't that great on performance and that people wanted to replace/remove it from the core. "On of the most interesting things about FreeSWITCH to me has been the fact that most data in the system such as registrations are kept in a SQL database. The default installation uses SQLite internally though you can easily point FreeSWITCH at one of a number of other SQL servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has become somewhat of a bottleneck in the core so future versions of FreeSWITCH will use less of it. For example, doing a "show channels" with over 500 channels in use starts to show issues. While I'm sad to see SQLite go in these cases, I am anxious to see how Minessale replaces it." http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale I was just being curious about it :-) Regards, Diego On Fri, Mar 13, 2009 at 10:09 AM, Mathieu Rene wrote: > In this case sqlite separates the core from what show channels uses > etc, the DB is filled in by an event handler. This actually increases > performance since you dont need to exclusively lock the session > manager's table just to list channels. > > > Math > > On 13-Mar-09, at 7:12 AM, Diego Viola wrote: > >> Erhm, this is what I wanted to say. >> >> -- >> >> I think FreeSWITCH should leave that SQLite option as optional, I >> think it would be nicer if FS doesn't depend on any SQL at all, if you >> do a show channels right now with -nosql, this is what you will see: >> >> freeswitch at internal> show channels >> -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! >> >> freeswitch at internal> >> >> I think it would be nice if FS can show you the information on real >> time, and on run time, without any SQL at all, and leave UnixODBC & >> SQLite as optional options, I also think this will improve >> performance. >> >> On Fri, Mar 13, 2009 at 6:53 AM, Diego Viola >> wrote: >>> This is what I get when I do a show channels with -nosql. >>> >>> freeswitch at internal> show channels >>> -ERR SQL DISABLED NO CHANNEL DATA AVAILABLE! >>> >>> freeswitch at internal> >>> >>> It would be nice if it prints the channels on runtime without the >>> need/overheat of SQL. >>> >>> Regards, >>> >>> Diego >>> >>> On Fri, Mar 13, 2009 at 6:46 AM, Diego Viola >>> wrote: >>>> Hello, >>>> >>>> I was wondering if it is possible to start FreeSWITCH without any >>>> SQL >>>> at all, yeah I know there is -nosql but I still want to have all the >>>> information available at run time, ie: show channels, sofia status, >>>> sofia status profile , everything. >>>> >>>> Would that improve performance? I ask this because I heard that >>>> SQLite >>>> causes some performance issues and I heard it's being removed from >>>> the >>>> core, I also think that making that optional would be nice along >>>> with >>>> UnixODBC support for everything. >>>> >>>> I also got a few situations where I had to rm -f >>>> /usr/local/freeswitch/db/* because there were some data there that >>>> was >>>> not supposed to be, that's why I think real time data on run time >>>> would be better. >>>> >>>> Anyway, just some feedback/question I wanted to share, FS is >>>> awesome, >>>> keep up the great work. >>>> >>>> Regards, >>>> >>>> Diego >>>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Mar 13 12:02:42 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Mar 2009 14:02:42 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> Message-ID: <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> Since we added indexes to the SQLite DB its not so bad. /b On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > Oh, I thought that SQLite wasn't that great on performance and that > people wanted to replace/remove it from the core. > > "On of the most interesting things about FreeSWITCH to me has been the > fact that most data in the system such as registrations are kept in a > SQL database. The default installation uses SQLite internally though > you can easily point FreeSWITCH at one of a number of other SQL > servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has > become somewhat of a bottleneck in the core so future versions of > FreeSWITCH will use less of it. For example, doing a "show channels" > with over 500 channels in use starts to show issues. While I'm sad to > see SQLite go in these cases, I am anxious to see how Minessale > replaces it." > > http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > > I was just being curious about it :-) > > Regards, > > Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090313/79793daf/attachment-0002.html From Prometheus001 at gmx.net Fri Mar 13 12:13:47 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 13 Mar 2009 20:13:47 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: References: Message-ID: <49BAB06B.2060009@gmx.net> Ok, I tried it. It works fine on dialthru. SDP header is enhanced. Perfect! However I have a problem which I already discussed before in this round, and I am very close to solve it, I think. Scenario is as follows * A callback application which is done through event socket calls party A with cid-number 'unknown' * Party A enters Digits which represents the number of party B * I set the caller-id-no and effective-caller-id-no to the value of A and sip_cid_type=rpid * Call is then transferred from A to Party B via the xml dialplan * Now I have 2 different variables when I trace the event_socket o caller-caller-id-number is 'anonymous' and this is submitted in the INVITE message to B o variable-caller-id-number is the right number of B, this is not part of the Sip message * B sees 'unknown' as the number of party A. This is also represented in the SIP mssage sent to B * I want B to see the right number of party A My question: What can I do to set to in the XML dialplan. Here is the part of the dialplan where I want so set it: Anybody has a solution? Best regards Peter Ken Rice schrieb: > These should be available any time you are going to process a call thru the > dialplan and call a bridge on the call > > > >> From: Peter P GMX >> Reply-To: >> Date: Tue, 10 Mar 2009 23:20:39 +0100 >> To: >> Subject: Re: [Freeswitch-users] Rewriting Remote Party ID >> >> Hello, >> >> are these variables only available at call setup time or can they be >> changed during a call, e.g. before a call is being transferred to >> another destination? >> >> Best regards >> Peter >> >> Michael Collins schrieb: >> >>> On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale >>> wrote: >>> >>> >>>> Latest SVN: >>>> >>>> Send no extra caller id info: >>>> {sip_cid_type=none}sofia/default/user at example.com >>>> >>>> Send RPID (default) >>>> {sip_cid_type=rpid}sofia/default/user at example.com >>>> >>>> Send P-XXX-Identity >>>> {sip_cid_type=pid}sofia/default/user at example.com >>>> >>>> Send RPID with chosen content >>>> {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_num >>>> ber=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul >>>> tuser at example.com >>>> >>>> >>> FYI, I added this info to the channel variables page: >>> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type >>> >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Fri Mar 13 12:57:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Mar 2009 12:57:17 -0700 Subject: [Freeswitch-users] New Voice Prompts - Need Community's Help Message-ID: <87f2f3b90903131257x78e763fdw55015566711cbb60@mail.gmail.com> FYI, I've just started a new JIRA for the new voice prompts list that we are getting together: http://jira.freeswitch.org/browse/FSSCRIPTS-15 Please add comments to this JIRA if you can think of useful prompts for FreeSWITCH that we don't already have recorded. We also need financial support to get these recorded. All those who wish to assist, please donate to brian at freeswitch.org on Paypal. Thanks again to all of you who make FreeSWITCH such a great community! -MC From gkuri at ieee.org Fri Mar 13 13:36:58 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 13 Mar 2009 13:36:58 -0700 Subject: [Freeswitch-users] inband dtmf detection Message-ID: <49BAC3EA.5030404@ieee.org> shoot me if I'm on the wrong track, but is it possible to use the start_dtmf app to do inband dtmf detection and "convert" the inband DTMF to rfc2833, as opposed to using the dtmf detection on a Linksys or Grandstream ATA? the reason I ask is the dtmf detection on these ATAs seems to falsely catch dtmf tones during certain conversations and falsely inject a dtmf tone into the stream (I believe the appropriate term is "talk off"). So I was thinking of setting the ATAs to do inband DTMF rather than the current rfc2833 setting, and having FS do the more reliable dtmf detection instead? Thanks, Gabe From brian at freeswitch.org Fri Mar 13 13:44:16 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Mar 2009 15:44:16 -0500 Subject: [Freeswitch-users] inband dtmf detection In-Reply-To: <49BAC3EA.5030404@ieee.org> References: <49BAC3EA.5030404@ieee.org> Message-ID: <58A70382-5FE1-4C26-B8B3-F161967F39E6@freeswitch.org> That is the purpose of start_dtmf to detect it inband and convert it to 2833. Unless you disabled 2833. /b On Mar 13, 2009, at 3:36 PM, Gabriel Kuri wrote: > shoot me if I'm on the wrong track, but is it possible to use the > start_dtmf app to do inband dtmf detection and "convert" the inband > DTMF > to rfc2833, as opposed to using the dtmf detection on a Linksys or > Grandstream ATA? From gkuri at ieee.org Fri Mar 13 14:33:45 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 13 Mar 2009 14:33:45 -0700 Subject: [Freeswitch-users] inband dtmf detection In-Reply-To: <58A70382-5FE1-4C26-B8B3-F161967F39E6@freeswitch.org> References: <49BAC3EA.5030404@ieee.org> <58A70382-5FE1-4C26-B8B3-F161967F39E6@freeswitch.org> Message-ID: <49BAD139.50907@ieee.org> thanks, I'll give it a try. I'm assuming this app requires a reasonable amount of additional processing power, particularly on a box with several hundred active channels, as it's basically snooping the rtp stream looking for inband dtmf tones? Gabe Brian West wrote: > That is the purpose of start_dtmf to detect it inband and convert it > to 2833. Unless you disabled 2833. > > /b > > On Mar 13, 2009, at 3:36 PM, Gabriel Kuri wrote: > >> shoot me if I'm on the wrong track, but is it possible to use the >> start_dtmf app to do inband dtmf detection and "convert" the inband >> DTMF >> to rfc2833, as opposed to using the dtmf detection on a Linksys or >> Grandstream ATA? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mszlazak at aol.com Fri Mar 13 16:38:28 2009 From: mszlazak at aol.com (mszlazak) Date: Fri, 13 Mar 2009 16:38:28 -0700 (PDT) Subject: [Freeswitch-users] Help with detecting hangup events. Message-ID: <22507114.post@talk.nabble.com> I have a problem in setting up an extension that will detect different hangup events. If an outside call comes in and gets sent to "myExtension" then that call is bridged to an application. When the application hangs up then the hangup hook "notify.js" is used to originate a call to another phone which notifies that phone that the application has been used. However, the problem here is that in the initial call, the caller can't hangup because then notify.js won't execute. I haven't tried this but if I changed things so "hang_after_bridge=false" then I'm guessing that notify.js will execute when the initial caller hangs up their phone. This I want but I don't want the call originated in notify.js to happen unless the bridged to application also issued a hangup previously. Otherwise, if the caller hangs up before the application does it's thing and hasn't issued a hangup then notify.js will still originate calls and these I don't want. So, I want notify.js to originate calls after the caller hangs up but these hangups have to be those that came after the bridged to application issued a hung up. I hope this makes sense. Thanks. Mark. -- View this message in context: http://www.nabble.com/Help-with-detecting-hangup-events.-tp22507114p22507114.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From diego.viola at gmail.com Fri Mar 13 21:12:28 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 14 Mar 2009 00:12:28 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> Message-ID: <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> Yeah, but still, it would be nice to see the channels with -nosql :) I don't want to be a pain in the ass, just giving some user feedback. Regards, Diego On Fri, Mar 13, 2009 at 3:02 PM, Brian West wrote: > Since we added indexes to the SQLite DB its not so bad. > /b > On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > > Oh, I thought that SQLite wasn't that great on performance and that > people wanted to replace/remove it from the core. > > "On of the most interesting things about FreeSWITCH to me has been the > fact that most data in the system such as registrations are kept in a > SQL database. The default installation uses SQLite internally though > you can easily point FreeSWITCH at one of a number of other SQL > servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has > become somewhat of a bottleneck in the core so future versions of > FreeSWITCH will use less of it. For example, doing a "show channels" > with over 500 channels in use starts to show issues. While I'm sad to > see SQLite go in these cases, I am anxious to see how Minessale > replaces it." > > http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > > I was just being curious about it :-) > > Regards, > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From yudha2008 at gmail.com Sat Mar 14 00:42:48 2009 From: yudha2008 at gmail.com (Baskar) Date: Sat, 14 Mar 2009 13:12:48 +0530 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: <49BA1D6F.30002@gmail.com> References: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> <49BA1D6F.30002@gmail.com> Message-ID: *Hi, I am using TDMAPI with EuroISDN. * *My openzap.conf* *[span wanpipe] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 [span wanpipe] name => OpenZAP number => 2 trunk_type => E1 b-channel => 2:1-15 d-channel => 2:16 b-channel => 2:17-31 * *My openzap.con.xml* * ** ** * * -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090314/0bbbb77d/attachment-0002.html From jp.manchu at gmail.com Sat Mar 14 04:49:02 2009 From: jp.manchu at gmail.com (JayaPrakash) Date: Sat, 14 Mar 2009 17:19:02 +0530 Subject: [Freeswitch-users] Issue relating mod_nibblebill Message-ID: Hi All, I am newbie to Freeswitch. I installed freeswitch-1.0.3 in Debian machine. I am able to make call, check presence, retrieve CDRs. I followed the installation steps given in mod_nibblebill for rating. While, installing mysql-connector-odbc, it has thrown errors related to mysql-config file, that it does not exist. Coming to mysql, mysql-client-5 and mysql-server-5 are installed. So I installed libmyodbc which is used for the same functionality. Rest of the steps are done, as given in mod-nibblebill installation. When the freeradius server is restarted, it has given the following error. 2009-03-14 14:55:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-03-14 14:55:47 [CRIT] mod_nibblebill.c:233 load_config() Cannot connect to ODBC driver/database freeswitch (user: root / pass dev)! Will you please have a look in solving this issue ? , how the issue can be solved? Thanks & Regards, JP. From mike at jerris.com Sat Mar 14 06:20:51 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Mar 2009 09:20:51 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> Message-ID: To clarify, -nosql turns on and off only the collecting of data for the show commands, and can now handle higher load than the sip stack can. The only thing your doing by saying -nosql is turning off the exact functionality you say you want. Its similar to saying I would like to support sip but don't want to load mod_sofia. There should be no reasons to use that command anymore, if you encounter any I would be interested in knowing what is going on. Mike On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > Yeah, but still, it would be nice to see the channels with -nosql :) > > I don't want to be a pain in the ass, just giving some user feedback. > > Regards, > > Diego > > On Fri, Mar 13, 2009 at 3:02 PM, Brian West > wrote: >> Since we added indexes to the SQLite DB its not so bad. >> /b >> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >> >> Oh, I thought that SQLite wasn't that great on performance and that >> people wanted to replace/remove it from the core. >> >> "On of the most interesting things about FreeSWITCH to me has been >> the >> fact that most data in the system such as registrations are kept in a >> SQL database. The default installation uses SQLite internally though >> you can easily point FreeSWITCH at one of a number of other SQL >> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >> become somewhat of a bottleneck in the core so future versions of >> FreeSWITCH will use less of it. For example, doing a "show channels" >> with over 500 channels in use starts to show issues. While I'm sad to >> see SQLite go in these cases, I am anxious to see how Minessale >> replaces it." >> >> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >> >> I was just being curious about it :-) >> >> Regards, >> >> Diego >> From Prometheus001 at gmx.net Sat Mar 14 06:29:40 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 14 Mar 2009 14:29:40 +0100 Subject: [Freeswitch-users] openzap with A102 In-Reply-To: References: <87f2f3b90903120957v338af9ady1713712a081a5e1c@mail.gmail.com> <49BA1D6F.30002@gmail.com> Message-ID: <49BBB144.60300@gmx.net> Hello Baskar, I had the same once. Please use in your configs. Euro dialect did not work any more for me in newer versions of freeswitch. Best regards Peter Baskar schrieb: > *Hi, > > I am using TDMAPI with EuroISDN. > * > *My openzap.conf* > > *[span wanpipe] > name => OpenZAP > number => 1 > trunk_type => E1 > b-channel => 1:1-15 > d-channel => 1:16 > b-channel => 1:17-31 > > [span wanpipe] > name => OpenZAP > number => 2 > trunk_type => E1 > b-channel => 2:1-15 > d-channel => 2:16 > b-channel => 2:17-31 > * > > *My openzap.con.xml* > * > > > > > > > > > > > > ** > > > > > ** > > * > * > > -- > Warm Regards, > N.Baskar > * > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Sat Mar 14 06:31:15 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 14 Mar 2009 14:31:15 +0100 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49BAB06B.2060009@gmx.net> References: <49BAB06B.2060009@gmx.net> Message-ID: <49BBB1A3.5010708@gmx.net> I got it to work. It was just so easy: Best regards Peter Peter P GMX schrieb: > Ok, I tried it. It works fine on dialthru. SDP header is enhanced. Perfect! > > However I have a problem which I already discussed before in this round, > and I am very close to solve it, I think. > Scenario is as follows > > * A callback application which is done through event socket calls > party A with cid-number 'unknown' > * Party A enters Digits which represents the number of party B > * I set the caller-id-no and effective-caller-id-no to the value of > A and sip_cid_type=rpid > * Call is then transferred from A to Party B via the xml dialplan > * Now I have 2 different variables when I trace the event_socket > o caller-caller-id-number is 'anonymous' and this is submitted > in the INVITE message to B > o variable-caller-id-number is the right number of B, this is > not part of the Sip message > * B sees 'unknown' as the number of party A. This is also > represented in the SIP mssage sent to B > * I want B to see the right number of party A > > My question: What can I do to set to > in the XML dialplan. > > Here is the part of the dialplan where I want so set it: > > > > > > > > > > Anybody has a solution? > > Best regards > Peter > > > > > Ken Rice schrieb: > >> These should be available any time you are going to process a call thru the >> dialplan and call a bridge on the call >> >> >> >> >>> From: Peter P GMX >>> Reply-To: >>> Date: Tue, 10 Mar 2009 23:20:39 +0100 >>> To: >>> Subject: Re: [Freeswitch-users] Rewriting Remote Party ID >>> >>> Hello, >>> >>> are these variables only available at call setup time or can they be >>> changed during a call, e.g. before a call is being transferred to >>> another destination? >>> >>> Best regards >>> Peter >>> >>> Michael Collins schrieb: >>> >>> >>>> On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale >>>> wrote: >>>> >>>> >>>> >>>>> Latest SVN: >>>>> >>>>> Send no extra caller id info: >>>>> {sip_cid_type=none}sofia/default/user at example.com >>>>> >>>>> Send RPID (default) >>>>> {sip_cid_type=rpid}sofia/default/user at example.com >>>>> >>>>> Send P-XXX-Identity >>>>> {sip_cid_type=pid}sofia/default/user at example.com >>>>> >>>>> Send RPID with chosen content >>>>> {sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_num >>>>> ber=1234,origination_privacy=screen+hide_name+hide_number}sofia/defaul >>>>> tuser at example.com >>>>> >>>>> >>>>> >>>> FYI, I added this info to the channel variables page: >>>> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type >>>> >>>> -MC >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From rupa at rupa.com Sat Mar 14 07:08:24 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Sat, 14 Mar 2009 09:08:24 -0500 Subject: [Freeswitch-users] Issue relating mod_nibblebill In-Reply-To: References: Message-ID: Have you configured unixodbc? http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc covers configuring odbc in detail. On Sat, Mar 14, 2009 at 6:49 AM, JayaPrakash wrote: > 2009-03-14 14:55:47 [ERR] switch_odbc.c:164 > switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: > [unixODBC][Driver Manager]Data source name not found, and no default > driver specified > This means unixODBC couldn't find the datasource you specified in it's configuration file. -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090314/50d6b70d/attachment-0002.html From anthony.minessale at gmail.com Sat Mar 14 07:15:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Mar 2009 09:15:46 -0500 Subject: [Freeswitch-users] Rewriting Remote Party ID In-Reply-To: <49B8E0AC.6070307@laposte.net> References: <49AFC1C3.9030603@laposte.net> <191c3a030903100616g16998df6k61937b67c79897bd@mail.gmail.com> <49B67984.30104@laposte.net> <191c3a030903100817xbfecc4fwc4bb33b59a0d19c6@mail.gmail.com> <49B7775D.3030308@laposte.net> <191c3a030903110616w6621230ege10039145492ba61@mail.gmail.com> <49B7C61A.6070201@laposte.net> <191c3a030903110814v4a550d65t107700cc178f3dd3@mail.gmail.com> <191c3a030903110815t64917635ma14aaa5c56fc72a0@mail.gmail.com> <49B8E0AC.6070307@laposte.net> Message-ID: <191c3a030903140715g3a77571bi45268ee36d632771@mail.gmail.com> origination_privacy was indeed broken fixed in r12603 On Thu, Mar 12, 2009 at 5:15 AM, rod wrote: > Thanks a lot Anthony, > > it's working great. > I'm just checking the origination_privacy parameter, cause it seems to > do nothing in my setup. > > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090314/a0ae9dac/attachment-0002.html From tayeb.meftah at gmail.com Sat Mar 14 09:47:36 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sat, 14 Mar 2009 17:47:36 +0100 Subject: [Freeswitch-users] Getting a free Did number for my FS Message-ID: <49BBDFA8.5010805@gmail.com> hello, please ho to get a free did number ? also, is it pocible to link it to my FS ? thanks From mszlazak at aol.com Sat Mar 14 10:50:57 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 14 Mar 2009 13:50:57 -0400 Subject: [Freeswitch-users] Problem with detecting session.cause with Linksys adaptor Message-ID: <8CB72E4902A36F4-B4C-EEE@webmail-mh27.sysops.aol.com> Hello, I have an analogue line connected to SPA3102 which then sends calls to my FreeSwitch (FS) application. I can originate a call from FS and dial out the analogue line almost fine. However, an issue seems to be detecting if the PSTN line is busy. Another issue might be "hanging up" the PSTN side of the adaptor from FS side. Anyway here's the problem. If I then try originating an outbound call from FS through the pstn side of the adaptor and if a caller on the analogue line has not hung up then they hear FS dialing and any automated message that FS then sends. I can not seem to detect from FS if the PSTN line is busy for some reason with the following code. var s; while (tryCalling()) {} s.hangup(); exit(); function tryCalling() { ??? s = new Session("sofia/gateway/spa3102/14082031170 at 10.0.0.5:5061"); ??? s.waitForAnswer(30000); ??? ??? if (s.cause == "USER_BUSY") { ??? ??? return true; ??? } ??? if (s.ready()) { ??? ??? s.sleep(1000); ??? ??? s.speak("cepstral","Callie","Hello from Gino Mick Gelato"); ??? } ??? return false; } I've also tried the above with ignore_early_media=true but no luck. Thanks. Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090314/0c3aabb9/attachment-0002.html From freeswitch at servercorps.com Sat Mar 14 10:54:34 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Sat, 14 Mar 2009 12:54:34 -0500 Subject: [Freeswitch-users] Getting a free Did number for my FS In-Reply-To: <49BBDFA8.5010805@gmail.com> References: <49BBDFA8.5010805@gmail.com> Message-ID: <92e7d2090903141054l5e090fa2vf51b423933872f22@mail.gmail.com> http://tinyurl.com/bohenh On Sat, Mar 14, 2009 at 11:47 AM, Meftah Tayeb wrote: > hello, > please ho to get a free did number ? > also, is it pocible to link it to my FS ? > thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From diego.viola at gmail.com Sat Mar 14 17:29:47 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 14 Mar 2009 20:29:47 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> Message-ID: <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> So how Asterisk does that "show channels" without SQL? I don't think they use SQLite internally. Just being curious. Diego On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > To clarify, -nosql turns on and off only the collecting of data for > the show commands, and can now handle higher load than the sip stack > can. ?The only thing your doing by saying -nosql is turning off the > exact functionality you say you want. ?Its similar to saying I would > like to support sip but don't want to load mod_sofia. ?There should be > no reasons to use that command anymore, if you encounter any I would > be interested in knowing what is going on. > > Mike > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > >> Yeah, but still, it would be nice to see the channels with -nosql :) >> >> I don't want to be a pain in the ass, just giving some user feedback. >> >> Regards, >> >> Diego >> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >> wrote: >>> Since we added indexes to the SQLite DB its not so bad. >>> /b >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >>> >>> Oh, I thought that SQLite wasn't that great on performance and that >>> people wanted to replace/remove it from the core. >>> >>> "On of the most interesting things about FreeSWITCH to me has been >>> the >>> fact that most data in the system such as registrations are kept in a >>> SQL database. The default installation uses SQLite internally though >>> you can easily point FreeSWITCH at one of a number of other SQL >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >>> become somewhat of a bottleneck in the core so future versions of >>> FreeSWITCH will use less of it. For example, doing a "show channels" >>> with over 500 channels in use starts to show issues. While I'm sad to >>> see SQLite go in these cases, I am anxious to see how Minessale >>> replaces it." >>> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >>> >>> I was just being curious about it :-) >>> >>> Regards, >>> >>> Diego >>> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Sat Mar 14 18:13:41 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Mar 2009 20:13:41 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> Message-ID: <019974CE-9251-4CA5-B7F9-B537C3447DFC@freeswitch.org> They walk across and lock every channel while printing the info... its a flawed way of doing things. /b On Mar 14, 2009, at 7:29 PM, Diego Viola wrote: > So how Asterisk does that "show channels" without SQL? I don't think > they use SQLite internally. > > Just being curious. > > Diego From anthony.minessale at gmail.com Sun Mar 15 06:30:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Mar 2009 08:30:24 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> Message-ID: <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> search google for bugs related to crash and show channels http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: > So how Asterisk does that "show channels" without SQL? I don't think > they use SQLite internally. > > Just being curious. > > Diego > > On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > > To clarify, -nosql turns on and off only the collecting of data for > > the show commands, and can now handle higher load than the sip stack > > can. The only thing your doing by saying -nosql is turning off the > > exact functionality you say you want. Its similar to saying I would > > like to support sip but don't want to load mod_sofia. There should be > > no reasons to use that command anymore, if you encounter any I would > > be interested in knowing what is going on. > > > > Mike > > > > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > > > >> Yeah, but still, it would be nice to see the channels with -nosql :) > >> > >> I don't want to be a pain in the ass, just giving some user feedback. > >> > >> Regards, > >> > >> Diego > >> > >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West > >> wrote: > >>> Since we added indexes to the SQLite DB its not so bad. > >>> /b > >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > >>> > >>> Oh, I thought that SQLite wasn't that great on performance and that > >>> people wanted to replace/remove it from the core. > >>> > >>> "On of the most interesting things about FreeSWITCH to me has been > >>> the > >>> fact that most data in the system such as registrations are kept in a > >>> SQL database. The default installation uses SQLite internally though > >>> you can easily point FreeSWITCH at one of a number of other SQL > >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has > >>> become somewhat of a bottleneck in the core so future versions of > >>> FreeSWITCH will use less of it. For example, doing a "show channels" > >>> with over 500 channels in use starts to show issues. While I'm sad to > >>> see SQLite go in these cases, I am anxious to see how Minessale > >>> replaces it." > >>> > >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > >>> > >>> I was just being curious about it :-) > >>> > >>> Regards, > >>> > >>> Diego > >>> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090315/de721507/attachment-0002.html From diego.viola at gmail.com Sun Mar 15 06:56:59 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 15 Mar 2009 09:56:59 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> Message-ID: <86a32abc0903150656w5d9f8a49t233e0cdda58482@mail.gmail.com> Wow, that sucks. It's clear now why it's done this way, keep up the great work. Diego On Sun, Mar 15, 2009 at 9:30 AM, Anthony Minessale wrote: > search google for bugs related to crash and show channels > http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a > > > On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: >> >> So how Asterisk does that "show channels" without SQL? I don't think >> they use SQLite internally. >> >> Just being curious. >> >> Diego >> >> On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: >> > To clarify, -nosql turns on and off only the collecting of data for >> > the show commands, and can now handle higher load than the sip stack >> > can. ?The only thing your doing by saying -nosql is turning off the >> > exact functionality you say you want. ?Its similar to saying I would >> > like to support sip but don't want to load mod_sofia. ?There should be >> > no reasons to use that command anymore, if you encounter any I would >> > be interested in knowing what is going on. >> > >> > Mike >> > >> > >> > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: >> > >> >> Yeah, but still, it would be nice to see the channels with -nosql :) >> >> >> >> I don't want to be a pain in the ass, just giving some user feedback. >> >> >> >> Regards, >> >> >> >> Diego >> >> >> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >> >> wrote: >> >>> Since we added indexes to the SQLite DB its not so bad. >> >>> /b >> >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >> >>> >> >>> Oh, I thought that SQLite wasn't that great on performance and that >> >>> people wanted to replace/remove it from the core. >> >>> >> >>> "On of the most interesting things about FreeSWITCH to me has been >> >>> the >> >>> fact that most data in the system such as registrations are kept in a >> >>> SQL database. The default installation uses SQLite internally though >> >>> you can easily point FreeSWITCH at one of a number of other SQL >> >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >> >>> become somewhat of a bottleneck in the core so future versions of >> >>> FreeSWITCH will use less of it. For example, doing a "show channels" >> >>> with over 500 channels in use starts to show issues. While I'm sad to >> >>> see SQLite go in these cases, I am anxious to see how Minessale >> >>> replaces it." >> >>> >> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >> >>> >> >>> I was just being curious about it :-) >> >>> >> >>> Regards, >> >>> >> >>> Diego >> >>> >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Sun Mar 15 07:16:05 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sun, 15 Mar 2009 10:16:05 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime In-Reply-To: <86a32abc0903150656w5d9f8a49t233e0cdda58482@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> <86a32abc0903150656w5d9f8a49t233e0cdda58482@mail.gmail.com> Message-ID: <86a32abc0903150716t6436e51kcf35f2c7a526df05@mail.gmail.com> Forget this, I don't want show channels without SQL anymore. Diego On Sun, Mar 15, 2009 at 9:56 AM, Diego Viola wrote: > Wow, that sucks. > > It's clear now why it's done this way, keep up the great work. > > Diego > > On Sun, Mar 15, 2009 at 9:30 AM, Anthony Minessale > wrote: >> search google for bugs related to crash and show channels >> http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a >> >> >> On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: >>> >>> So how Asterisk does that "show channels" without SQL? I don't think >>> they use SQLite internally. >>> >>> Just being curious. >>> >>> Diego >>> >>> On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: >>> > To clarify, -nosql turns on and off only the collecting of data for >>> > the show commands, and can now handle higher load than the sip stack >>> > can. ?The only thing your doing by saying -nosql is turning off the >>> > exact functionality you say you want. ?Its similar to saying I would >>> > like to support sip but don't want to load mod_sofia. ?There should be >>> > no reasons to use that command anymore, if you encounter any I would >>> > be interested in knowing what is going on. >>> > >>> > Mike >>> > >>> > >>> > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: >>> > >>> >> Yeah, but still, it would be nice to see the channels with -nosql :) >>> >> >>> >> I don't want to be a pain in the ass, just giving some user feedback. >>> >> >>> >> Regards, >>> >> >>> >> Diego >>> >> >>> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >>> >> wrote: >>> >>> Since we added indexes to the SQLite DB its not so bad. >>> >>> /b >>> >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >>> >>> >>> >>> Oh, I thought that SQLite wasn't that great on performance and that >>> >>> people wanted to replace/remove it from the core. >>> >>> >>> >>> "On of the most interesting things about FreeSWITCH to me has been >>> >>> the >>> >>> fact that most data in the system such as registrations are kept in a >>> >>> SQL database. The default installation uses SQLite internally though >>> >>> you can easily point FreeSWITCH at one of a number of other SQL >>> >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >>> >>> become somewhat of a bottleneck in the core so future versions of >>> >>> FreeSWITCH will use less of it. For example, doing a "show channels" >>> >>> with over 500 channels in use starts to show issues. While I'm sad to >>> >>> see SQLite go in these cases, I am anxious to see how Minessale >>> >>> replaces it." >>> >>> >>> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >>> >>> >>> >>> I was just being curious about it :-) >>> >>> >>> >>> Regards, >>> >>> >>> >>> Diego >>> >>> >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From marc at kasteris.com Sun Mar 15 10:19:13 2009 From: marc at kasteris.com (Marc Orenberg) Date: Sun, 15 Mar 2009 10:19:13 -0700 (PDT) Subject: [Freeswitch-users] Problems with mp3 file formats in mod_shout Message-ID: <276762.70160.qm@web50806.mail.re2.yahoo.com> Hi, I'm using mod_shout to play mp3 files, and I'm having?some trouble. I have two sets of mp3 files, one from a professional voice-prompt service, and one set that I converted from .wav files.? In both sets, all of the files play fine in Windows Media player and other players, but in both sets, there are some files which mod_shout is complaining about. It gives me errors like this: ??? ??? Note: Illegal Audio-MPEG-Header 0x41504554 at offset 0x40fa. ??? ??? Note: Trying to resync... ??? ??? Note: Hit end of (available) data during resync. When it gets these errors, it often stops playing the file before it is completed. I've tried checking / re-encoding these files using Audacity, vbrfix, mp3validator and mp3gain, and I continue to have the same problem, which is making me wonder if it's really a formating problem after all, or maybe some bug in mod_shout. I'm wondering if anybody has had these problems before, and if they have any insight into what could be going on. I'd really appreciate any help or advice. By the way, I'm using FreeSwitch version 1.0.3 on CentOS linux. Thanks in advance, Marc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090315/68619a88/attachment-0002.html From gcd at i.ph Sun Mar 15 16:46:09 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 16 Mar 2009 07:46:09 +0800 Subject: [Freeswitch-users] Getting a free Did number for my FS In-Reply-To: <49BBDFA8.5010805@gmail.com> References: <49BBDFA8.5010805@gmail.com> Message-ID: <7d0bfd8c0903151646p47700a12v1e9c601518fc8460@mail.gmail.com> perhaps you're referring to VPN (Virtual Phone Number). you can visit http://www.ipkall.com that offers free Washington state numbers. On Sun, Mar 15, 2009 at 12:47 AM, Meftah Tayeb wrote: > hello, > please ho to get a free did number ? > also, is it pocible to link it to my FS ? > thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/c51eca85/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sun Mar 15 17:01:59 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 16 Mar 2009 00:01:59 -0000 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at thesame time have all info available on realtime/runtime In-Reply-To: <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com><86a32abc0903130353n1d13965fm3efe3acf634006a7@mail.gmail.com><86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com><25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca><86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com><984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org><86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com><86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> Message-ID: To be fair, most of these messages are 4-5 years old. That said to date, I can crash * by repeatedly doing a 'show channels'. All the same FS should be robust enough to suffer this abuse. If it's not,. the issue needs to be investigated. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 15 March 2009 13:30 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but at thesame time have all info available on realtime/runtime search google for bugs related to crash and show channels http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe= utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: So how Asterisk does that "show channels" without SQL? I don't think they use SQLite internally. Just being curious. Diego On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > To clarify, -nosql turns on and off only the collecting of data for > the show commands, and can now handle higher load than the sip stack > can. The only thing your doing by saying -nosql is turning off the > exact functionality you say you want. Its similar to saying I would > like to support sip but don't want to load mod_sofia. There should be > no reasons to use that command anymore, if you encounter any I would > be interested in knowing what is going on. > > Mike > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > >> Yeah, but still, it would be nice to see the channels with -nosql :) >> >> I don't want to be a pain in the ass, just giving some user feedback. >> >> Regards, >> >> Diego >> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >> wrote: >>> Since we added indexes to the SQLite DB its not so bad. >>> /b >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >>> >>> Oh, I thought that SQLite wasn't that great on performance and that >>> people wanted to replace/remove it from the core. >>> >>> "On of the most interesting things about FreeSWITCH to me has been >>> the >>> fact that most data in the system such as registrations are kept in a >>> SQL database. The default installation uses SQLite internally though >>> you can easily point FreeSWITCH at one of a number of other SQL >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >>> become somewhat of a bottleneck in the core so future versions of >>> FreeSWITCH will use less of it. For example, doing a "show channels" >>> with over 500 channels in use starts to show issues. While I'm sad to >>> see SQLite go in these cases, I am anxious to see how Minessale >>> replaces it." >>> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >>> >>> I was just being curious about it :-) >>> >>> Regards, >>> >>> Diego >>> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/ca3a1b6b/attachment-0002.html From anthony.minessale at gmail.com Sun Mar 15 17:15:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Mar 2009 19:15:47 -0500 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but at thesame time have all info available on realtime/runtime In-Reply-To: References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com> <86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com> <25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca> <86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com> <984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org> <86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com> <86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com> <191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> Message-ID: <191c3a030903151715w34dc4d2el82331e77f3b558d6@mail.gmail.com> Perhaps a misunderstanding? We do not suffer from any problem at all regarding "show channels". The reason for the link was to demonstrate the issue we are familiar with from our asterisk days (3-4 years ago) and to help explain how we solved it by storing the calls states in a separate table to avoid locking the channels. You can type show channel in FS all you want and all you are doing is selecting from the SQLite db. The Original question by the poster was if we can find a way to turn off SQL and still allow show channels to work and the answer is, sorry no. On Sun, Mar 15, 2009 at 7:01 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > To be fair, most of these messages are 4-5 years old. That said to date, > I can crash * by repeatedly doing a ?show channels?. All the same FS should > be robust enough to suffer this abuse. If it?s not,. the issue needs to be > investigated. > > > > > > Regards, > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 15 March 2009 13:30 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Start FreeSWITCH without any SQL but at > thesame time have all info available on realtime/runtime > > > > search google for bugs related to crash and show channels > > http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a > > On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola > wrote: > > So how Asterisk does that "show channels" without SQL? I don't think > they use SQLite internally. > > Just being curious. > > Diego > > > On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > > To clarify, -nosql turns on and off only the collecting of data for > > the show commands, and can now handle higher load than the sip stack > > can. The only thing your doing by saying -nosql is turning off the > > exact functionality you say you want. Its similar to saying I would > > like to support sip but don't want to load mod_sofia. There should be > > no reasons to use that command anymore, if you encounter any I would > > be interested in knowing what is going on. > > > > Mike > > > > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > > > >> Yeah, but still, it would be nice to see the channels with -nosql :) > >> > >> I don't want to be a pain in the ass, just giving some user feedback. > >> > >> Regards, > >> > >> Diego > >> > >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West > >> wrote: > >>> Since we added indexes to the SQLite DB its not so bad. > >>> /b > >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > >>> > >>> Oh, I thought that SQLite wasn't that great on performance and that > >>> people wanted to replace/remove it from the core. > >>> > >>> "On of the most interesting things about FreeSWITCH to me has been > >>> the > >>> fact that most data in the system such as registrations are kept in a > >>> SQL database. The default installation uses SQLite internally though > >>> you can easily point FreeSWITCH at one of a number of other SQL > >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has > >>> become somewhat of a bottleneck in the core so future versions of > >>> FreeSWITCH will use less of it. For example, doing a "show channels" > >>> with over 500 channels in use starts to show issues. While I'm sad to > >>> see SQLite go in these cases, I am anxious to see how Minessale > >>> replaces it." > >>> > >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > >>> > >>> I was just being curious about it :-) > >>> > >>> Regards, > >>> > >>> Diego > >>> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090315/d09cb47e/attachment-0002.html From shanwlin at gmail.com Mon Mar 16 00:18:00 2009 From: shanwlin at gmail.com (shawn lin) Date: Mon, 16 Mar 2009 15:18:00 +0800 Subject: [Freeswitch-users] How can I change the timeout config? Message-ID: <9fcf45ed0903160018wad781f4k99e472b5170eb7f8@mail.gmail.com> Hi all, I met a problem which I thought is caused by the timeout config setting. I use SIPp UAC <-> FreeSWITCH <-> SIPp UAS to do my test. Commands: FS: ulimit -s 256; sudo ./freeswitch UAS: ulimit -s 256; sudo sipp -sn uas -i 10.67.7.224 10.67.7.46 -trace_err UAC: ulimit -s 256; sudo sipp -sn uac -i 10.67.7.213 10.67.7.29 -trace_err -l 5000 -m 5000 -d 300000 -r 5 Infr: UAC(10.67.7.213) <-> FS internal(10.67.7.29) FS external(10.67.7.46) <-> UAS 10.67.7.224 ********************** *** My problem **** ********************** I delayed uac for 300 seconds to send the BYE messages to uas, but after about 40 seconds of pausing, uas received BYE messages from FreeSWITCH: 2009-03-16 14:36:16:274 1237185376.274928: Aborting call on an unexpected BYE for call: 1-16380 at 10.67.7.213. **** Help *** Can anyone tell me how to release the timeout limit? Best Regards! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/7e26bd25/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Mar 16 04:52:59 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 16 Mar 2009 11:52:59 -0000 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but atthesame time have all info available on realtime/runtime In-Reply-To: <191c3a030903151715w34dc4d2el82331e77f3b558d6@mail.gmail.com> References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com><86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com><25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca><86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com><984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org><86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com><86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com><191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> <191c3a030903151715w34dc4d2el82331e77f3b558d6@mail.gmail.com> Message-ID: Yup, my mistake, got the wrong end of the stick. However while we're on the subject of show channels, is it possible to get a formatted response back from the event socket? It would be nice to interrogate an event style response. Perhaps even including some other goodies such as load etc. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 16 March 2009 00:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but atthesame time have all info available on realtime/runtime Perhaps a misunderstanding? We do not suffer from any problem at all regarding "show channels". The reason for the link was to demonstrate the issue we are familiar with from our asterisk days (3-4 years ago) and to help explain how we solved it by storing the calls states in a separate table to avoid locking the channels. You can type show channel in FS all you want and all you are doing is selecting from the SQLite db. The Original question by the poster was if we can find a way to turn off SQL and still allow show channels to work and the answer is, sorry no. On Sun, Mar 15, 2009 at 7:01 PM, Nik Middleton wrote: To be fair, most of these messages are 4-5 years old. That said to date, I can crash * by repeatedly doing a 'show channels'. All the same FS should be robust enough to suffer this abuse. If it's not,. the issue needs to be investigated. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 15 March 2009 13:30 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but at thesame time have all info available on realtime/runtime search google for bugs related to crash and show channels http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe= utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola wrote: So how Asterisk does that "show channels" without SQL? I don't think they use SQLite internally. Just being curious. Diego On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris wrote: > To clarify, -nosql turns on and off only the collecting of data for > the show commands, and can now handle higher load than the sip stack > can. The only thing your doing by saying -nosql is turning off the > exact functionality you say you want. Its similar to saying I would > like to support sip but don't want to load mod_sofia. There should be > no reasons to use that command anymore, if you encounter any I would > be interested in knowing what is going on. > > Mike > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > >> Yeah, but still, it would be nice to see the channels with -nosql :) >> >> I don't want to be a pain in the ass, just giving some user feedback. >> >> Regards, >> >> Diego >> >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West >> wrote: >>> Since we added indexes to the SQLite DB its not so bad. >>> /b >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: >>> >>> Oh, I thought that SQLite wasn't that great on performance and that >>> people wanted to replace/remove it from the core. >>> >>> "On of the most interesting things about FreeSWITCH to me has been >>> the >>> fact that most data in the system such as registrations are kept in a >>> SQL database. The default installation uses SQLite internally though >>> you can easily point FreeSWITCH at one of a number of other SQL >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite has >>> become somewhat of a bottleneck in the core so future versions of >>> FreeSWITCH will use less of it. For example, doing a "show channels" >>> with over 500 channels in use starts to show issues. While I'm sad to >>> see SQLite go in these cases, I am anxious to see how Minessale >>> replaces it." >>> >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale >>> >>> I was just being curious about it :-) >>> >>> Regards, >>> >>> Diego >>> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d528c13c/attachment-0002.html From mrene_lists at avgs.ca Mon Mar 16 04:55:14 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 16 Mar 2009 07:55:14 -0400 Subject: [Freeswitch-users] Start FreeSWITCH without any SQL but atthesame time have all info available on realtime/runtime In-Reply-To: References: <86a32abc0903130346y56352da7m95a9be0063901096@mail.gmail.com><86a32abc0903130412r59620f5rc8dd79d382bb68d7@mail.gmail.com><25442B9D-FBD9-4CE7-B88C-05FA68B26862@avgs.ca><86a32abc0903130736h33aefd77pe2416eed7b274b45@mail.gmail.com><984E251D-D9F1-4B41-9CD0-AE723A546DA4@freeswitch.org><86a32abc0903132112v24e155fej5ad7e0b3ef675915@mail.gmail.com><86a32abc0903141729k134db737w246d8678b96139c2@mail.gmail.com><191c3a030903150630q2d8c75adwd8061a8d35429409@mail.gmail.com> <191c3a030903151715w34dc4d2el82331e77f3b558d6@mail.gmail.com> Message-ID: <9256965F-58D6-48E1-A72E-31F13687FE26@avgs.ca> Depending on the API you're calling, of course show channels as xml show calls as xml uuid_dump [uuid] xml sofia xmlstatus and probably more than that. Math On 16-Mar-09, at 7:52 AM, Nik Middleton wrote: > Yup, my mistake, got the wrong end of the stick. > > However while we?re on the subject of show channels, is it possible > to get a formatted response back from the event socket? It would be > nice to interrogate an event style response. Perhaps even including > some other goodies such as load etc. > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Anthony Minessale > Sent: 16 March 2009 00:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but > atthesame time have all info available on realtime/runtime > > Perhaps a misunderstanding? > > We do not suffer from any problem at all regarding "show channels". > > The reason for the link was to demonstrate the issue we are familiar > with from our asterisk days (3-4 years ago) and to help explain > how we solved it by storing the calls states in a separate table to > avoid locking the channels. > > You can type show channel in FS all you want and all you are doing > is selecting from the SQLite db. > > The Original question by the poster was if we can find a way to turn > off SQL and still allow show channels to work and the > answer is, sorry no. > > > On Sun, Mar 15, 2009 at 7:01 PM, Nik Middleton > wrote: > To be fair, most of these messages are 4-5 years old. That said to > date, I can crash * by repeatedly doing a ?show channels?. All the > same FS should be robust enough to suffer this abuse. If it?s not,. > the issue needs to be investigated. > > > > > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Anthony Minessale > Sent: 15 March 2009 13:30 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Start FreeSWITCH without any SQL but > at thesame time have all info available on realtime/runtime > > > > search google for bugs related to crash and show channels > http://www.google.com/search?q=asterisk+crash+show+channels&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a > > On Sat, Mar 14, 2009 at 7:29 PM, Diego Viola > wrote: > > So how Asterisk does that "show channels" without SQL? I don't think > they use SQLite internally. > > Just being curious. > > Diego > > > On Sat, Mar 14, 2009 at 9:20 AM, Michael Jerris > wrote: > > To clarify, -nosql turns on and off only the collecting of data for > > the show commands, and can now handle higher load than the sip stack > > can. The only thing your doing by saying -nosql is turning off the > > exact functionality you say you want. Its similar to saying I would > > like to support sip but don't want to load mod_sofia. There > should be > > no reasons to use that command anymore, if you encounter any I would > > be interested in knowing what is going on. > > > > Mike > > > > > > On Mar 14, 2009, at 12:12 AM, Diego Viola wrote: > > > >> Yeah, but still, it would be nice to see the channels with - > nosql :) > >> > >> I don't want to be a pain in the ass, just giving some user > feedback. > >> > >> Regards, > >> > >> Diego > >> > >> On Fri, Mar 13, 2009 at 3:02 PM, Brian West > >> wrote: > >>> Since we added indexes to the SQLite DB its not so bad. > >>> /b > >>> On Mar 13, 2009, at 9:36 AM, Diego Viola wrote: > >>> > >>> Oh, I thought that SQLite wasn't that great on performance and > that > >>> people wanted to replace/remove it from the core. > >>> > >>> "On of the most interesting things about FreeSWITCH to me has been > >>> the > >>> fact that most data in the system such as registrations are kept > in a > >>> SQL database. The default installation uses SQLite internally > though > >>> you can easily point FreeSWITCH at one of a number of other SQL > >>> servers such as PostgreSQL or MySQL via UnixODBC. Sadly, SQLite > has > >>> become somewhat of a bottleneck in the core so future versions of > >>> FreeSWITCH will use less of it. For example, doing a "show > channels" > >>> with over 500 channels in use starts to show issues. While I'm > sad to > >>> see SQLite go in these cases, I am anxious to see how Minessale > >>> replaces it." > >>> > >>> http://www.anders.com/cms/275/FreeSWITCH/ClueCon/Anthony.Minessale > >>> > >>> I was just being curious about it :-) > >>> > >>> Regards, > >>> > >>> Diego > >>> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/701f2305/attachment-0002.html From keithl at voxtelecom.co.za Mon Mar 16 06:36:01 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Mon, 16 Mar 2009 15:36:01 +0200 Subject: [Freeswitch-users] Core Dump on receiving a call from device with 'broken' G.722 codec. Message-ID: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> Hi, I am on fs 1.0.trunk (12530M) testing G.722 and found that when using a 'broken' configuration from a softphone configured for G.722, I get the warning on the cli: "We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. " when fs gets the invite, but then does a core dump when it tries to: Below are some of the traces and info output from before the core dump happens. I see this when I run gdb on the dumpfile. #0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/8154%172.16.1.3", timelimit_sec=30, table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=) at src/switch_ivr_originate.c:1609 1609 if (switch_core_codec_init(&write_codec, (gdb) frame 1 #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so I wonder if anybody else has seen this behavior? This happens when the destination phone is also G.722 capable (policom). If I change the "frame per packet" setting in the softphone to 2 - All works OK (but the default is 1 - so cant risk allowing G.722 if it's going to core dump fs if a user make a wrong configuration) Best Regards Keith ************************************************************************ ************************************************************* 2009-03-16 14:37:58 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/sprof1/27879998182 at 196.99.88.77 [47e0c972-1227-11de-8b8e-1789e43c417d] <.....> 2009-03-16 14:37:58 [INFO] mod_sofia.c:1310 sofia_receive_message() Asked to send early media by sofia/sprof1/27879998182 at 196.99.88.77 2009-03-16 14:37:58 [NOTICE] sofia_glue.c:2245 sofia_glue_tech_media() Pre-Answer sofia/sprof1/27879998182 at 196.99.88.77! 2009-03-16 14:37:58 [INFO] mod_sofia.c:1351 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1237190146 1237190147 IN IP4 196.99.88.77 s=FreeSWITCH c=IN IP4 196.99.88.77 t=0 0 m=audio 16932 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-03-16 14:37:58 [INFO] switch_rtp.c:1441 rtp_common_read() Auto Changing port from 172.16.0.63:29081 to 196.22.33.44:10634 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() -- checktalktime.js -- <.. In this js I do http call to collect maximum talktime allowed ..> 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() schedparms=+3600 tbhangupwarn XML hangupwarn 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182 at 196.99.88.77 to XML[27879998154 at e164route] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->27879998154 in context e164route 2009-03-16 14:37:59 [INFO] mod_dptools.c:945 info_function() CHANNEL_DATA: Event-Name: [CHANNEL_DATA] Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] FreeSWITCH-Hostname: [myfsbox] FreeSWITCH-IPv4: [196.99.88.77] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-03-16 14:37:59] Event-Date-GMT: [Mon, 16 Mar 2009 12:37:59 GMT] Event-Date-Timestamp: [1237207079387869] Event-Calling-File: [mod_dptools.c] Event-Calling-Function: [info_function] Event-Calling-Line-Number: [941] Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/sprof1/27879998182 at 196.99.88.77] Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [early] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Caller-Username: [27879998182] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MeMe] Caller-Caller-ID-Number: [27879998182] Caller-Network-Addr: [196.22.33.44] Caller-Destination-Number: [27879998154] Caller-Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Caller-Source: [mod_sofia] Caller-Context: [e164route] Caller-RDNIS: [27879998154] Caller-Channel-Name: [sofia/sprof1/27879998182 at 196.99.88.77] Caller-Profile-Index: [4] Caller-Profile-Created-Time: [1237207079387869] Caller-Channel-Created-Time: [1237207078659653] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [1237207078679638] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [196.22.33.44] variable_sip_received_port: [36745] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_Event-Name: [REQUEST_PARAMS] variable_Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] variable_FreeSWITCH-Hostname: [myfsbox] variable_FreeSWITCH-IPv4: [196.99.88.77] variable_FreeSWITCH-IPv6: [::1] variable_Event-Date-Local: [2009-03-16 14:37:58] variable_Event-Date-GMT: [Mon, 16 Mar 2009 12:37:58 GMT] variable_Event-Date-Timestamp: [1237207078659653] variable_Event-Calling-File: [sofia_reg.c] variable_Event-Calling-Function: [sofia_reg_parse_auth] variable_Event-Calling-Line-Number: [1727] variable_sip_mailbox: [879998182] variable_sip_auth_username: [27879998182] variable_sip_auth_realm: [196.99.88.77] variable_mailbox: [879998182] variable_user_name: [27879998182] variable_domain_name: [196.99.88.77] variable_record_stereo: [true] variable_default_gateway: [verso] variable_default_areacode: [87] variable_transfer_fallback_extension: [operator] variable_sip-force-expires: [180] variable_toll_allow: [domestic,international] variable_accountcode: [tbaaaa] variable_user_context: [sprof1] variable_effective_caller_id_name: [TickyBox 99999 Phone 99 Test] variable_effective_caller_id_number: [879998182] variable_outbound_caller_id_name: [879998150] variable_outbound_caller_id_number: [879998150] variable_sip_from_user: [27879998182] variable_sip_from_uri: [27879998182 at 196.99.88.77] variable_sip_from_host: [196.99.88.77] variable_sip_from_user_stripped: [27879998182] variable_sip_from_tag: [196.99.88.77] variable_sofia_profile_name: [sprof1] variable_sofia_profile_domain_name: [196.99.88.77] variable_sip_req_user: [0879998154] variable_sip_req_uri: [0879998154 at 196.99.88.77] variable_sip_req_host: [196.99.88.77] variable_sip_to_user: [0879998154] variable_sip_to_uri: [0879998154 at 196.99.88.77] variable_sip_to_host: [196.99.88.77] variable_sip_contact_user: [27879998182] variable_sip_contact_port: [22034] variable_sip_contact_uri: [27879998182 at 172.16.0.63:22034] variable_sip_contact_host: [172.16.0.63] variable_channel_name: [sofia/sprof1/27879998182 at 196.99.88.77] variable_sip_call_id: [xr125298731411533c30039109e1921f at 192.168.10.1] variable_sip_user_agent: [BrokenPhone/1.4.2] variable_sip_via_host: [172.16.0.63] variable_sip_via_port: [22034] variable_sip_via_rport: [36745] variable_presence_id: [27879998182 at 196.99.88.77] variable_switch_r_sdp: [v=0 o=2787999818 2265 2267 IN IP4 172.16.0.63 s=Broken c=IN IP4 172.16.0.63 t=0 0 m=audio 29081 RTP/AVP 9 18 101 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=candidates:-1564465265,172.16.0.63:29081,192.168.10.1:29081,192.168.20 .1:29081 ] variable_outboundcontext: [setupprepaycall] variable_remote_media_ip: [172.16.0.63] variable_remote_media_port: [29081] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_local_media_ip: [196.99.88.77] variable_local_media_port: [16932] variable_endpoint_disposition: [EARLY MEDIA] variable_sip_nat_detected: [true] variable_api_hangup_hook: [jsapi::completecall.js] variable_talktime: [6870] variable_action: [allow] variable_status: [allowed] variable_integer: [102] variable_fraction: [85] variable_saytalktime: [60:0] variable_schedparms: [+3600 tbhangupwarn XML hangupwarn] variable_bridgejscb: [{api_hangup_hook=jsapi::completecall.js}] variable_max_forwards: [67] variable_current_application: [info] 2009-03-16 14:37:59 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 820 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-03-16 14:37:59 [WARNING] switch_core_codec.c:499 switch_core_codec_init() Codec G722 Exists but not at the desired implementation. 8000hz 820ms 2009-03-16 14:37:59 [ERR] sofia_glue.c:1700 sofia_glue_tech_set_codec() Can't load codec? 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/sprof1/27879998182 at 196.99.88.77 has no read codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:585 switch_core_session_write_frame() sofia/sprof1/27879998182 at 196.99.88.77 has no write codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/sprof1/27879998182 at 196.99.88.77 has no read codec. 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182 at 196.99.88.77 to XML[8154 at toregext] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->8154 in context toregext 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/27879998182.2009-03-16- 14-37-59.wav 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features <... output from info application ...> 2009-03-16 14:38:00 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/voxwan/8154 [48c2f400-1227-11de-8b8e-1789e43c417d] Segmentation fault (core dumped) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/7b37f236/attachment-0002.html From mrene_lists at avgs.ca Mon Mar 16 06:46:08 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 16 Mar 2009 09:46:08 -0400 Subject: [Freeswitch-users] Core Dump on receiving a call from device with 'broken' G.722 codec. In-Reply-To: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> Message-ID: Hi, You should be reporting this on JIRA ( see http://wiki.freeswitch.org/wiki/Reporting_Bugs ) Also please include a "bt", not just "frame 1" as it doesnt give out much information. Math On 16-Mar-09, at 9:36 AM, Keith Laaks wrote: > Hi, > > I am on fs 1.0.trunk (12530M) testing G.722 and found that when > using a ?broken? configuration from a softphone configured for G. > 722, I get the warning on the cli: > > ?We were told to use ptime 20 but what they meant to say was 820 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sipura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who knows what will happen.. > ? > when fs gets the invite, but then does a core dump when it tries to: > > > > > Below are some of the traces and info output from before the core > dump happens. > > I see this when I run gdb on the dumpfile. > > #0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, > bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/ > 8154%172.16.1.3", timelimit_sec=30, > table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, > caller_profile_override=0x0, ovars=0x0, flags=) > at src/switch_ivr_originate.c:1609 > 1609 if > (switch_core_codec_init(&write_codec, > > (gdb) frame 1 > #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so > > > I wonder if anybody else has seen this behavior? > > This happens when the destination phone is also G.722 capable > (policom). > If I change the ?frame per packet? setting in the softphone to 2 ? > All works OK (but the default is 1 ? so cant risk allowing G.722 if > it?s going to core dump fs if a user make a wrong configuration) > > > Best Regards > > Keith > > > ************************************************************************************************************************************* > > > 2009-03-16 14:37:58 [NOTICE] switch_channel.c:592 > switch_channel_set_name() New Channel sofia/sprof1/27879998182 at 196.99.88.77 > [47e0c972-1227-11de-8b8e-1789e43c417d] > > <.....> > > 2009-03-16 14:37:58 [INFO] mod_sofia.c:1310 sofia_receive_message() > Asked to send early media by sofia/sprof1/27879998182 at 196.99.88.77 > 2009-03-16 14:37:58 [NOTICE] sofia_glue.c:2245 > sofia_glue_tech_media() Pre-Answer sofia/sprof1/27879998182 at 196.99.88.77 > ! > 2009-03-16 14:37:58 [INFO] mod_sofia.c:1351 sofia_receive_message() > Ring SDP: > v=0 > o=FreeSWITCH 1237190146 1237190147 IN IP4 196.99.88.77 > s=FreeSWITCH > c=IN IP4 196.99.88.77 > t=0 0 > m=audio 16932 RTP/AVP 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2009-03-16 14:37:58 [INFO] switch_rtp.c:1441 rtp_common_read() Auto > Changing port from 172.16.0.63:29081 to 196.22.33.44:10634 > > 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() -- > checktalktime.js -- > > <.. In this js I do http call to collect maximum talktime allowed ..> > > 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() > schedparms=+3600 tbhangupwarn XML hangupwarn > 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 > switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182 at 196.99.88.77 > to XML[27879998154 at e164route] > 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing MeMe->27879998154 in context e164route > 2009-03-16 14:37:59 [INFO] mod_dptools.c:945 info_function() > CHANNEL_DATA: > Event-Name: [CHANNEL_DATA] > Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] > FreeSWITCH-Hostname: [myfsbox] > FreeSWITCH-IPv4: [196.99.88.77] > FreeSWITCH-IPv6: [::1] > Event-Date-Local: [2009-03-16 14:37:59] > Event-Date-GMT: [Mon, 16 Mar 2009 12:37:59 GMT] > Event-Date-Timestamp: [1237207079387869] > Event-Calling-File: [mod_dptools.c] > Event-Calling-Function: [info_function] > Event-Calling-Line-Number: [941] > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/sprof1/27879998182 at 196.99.88.77] > Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [early] > Channel-Read-Codec-Name: [G722] > Channel-Read-Codec-Rate: [16000] > Channel-Write-Codec-Name: [G722] > Channel-Write-Codec-Rate: [16000] > Caller-Username: [27879998182] > Caller-Dialplan: [XML] > Caller-Caller-ID-Name: [MeMe] > Caller-Caller-ID-Number: [27879998182] > Caller-Network-Addr: [196.22.33.44] > Caller-Destination-Number: [27879998154] > Caller-Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] > Caller-Source: [mod_sofia] > Caller-Context: [e164route] > Caller-RDNIS: [27879998154] > Caller-Channel-Name: [sofia/sprof1/27879998182 at 196.99.88.77] > Caller-Profile-Index: [4] > Caller-Profile-Created-Time: [1237207079387869] > Caller-Channel-Created-Time: [1237207078659653] > Caller-Channel-Answered-Time: [0] > Caller-Channel-Progress-Time: [0] > Caller-Channel-Progress-Media-Time: [1237207078679638] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > variable_sip_received_ip: [196.22.33.44] > variable_sip_received_port: [36745] > variable_sip_via_protocol: [udp] > variable_sip_authorized: [true] > variable_Event-Name: [REQUEST_PARAMS] > variable_Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] > variable_FreeSWITCH-Hostname: [myfsbox] > variable_FreeSWITCH-IPv4: [196.99.88.77] > variable_FreeSWITCH-IPv6: [::1] > variable_Event-Date-Local: [2009-03-16 14:37:58] > variable_Event-Date-GMT: [Mon, 16 Mar 2009 12:37:58 GMT] > variable_Event-Date-Timestamp: [1237207078659653] > variable_Event-Calling-File: [sofia_reg.c] > variable_Event-Calling-Function: [sofia_reg_parse_auth] > variable_Event-Calling-Line-Number: [1727] > variable_sip_mailbox: [879998182] > variable_sip_auth_username: [27879998182] > variable_sip_auth_realm: [196.99.88.77] > variable_mailbox: [879998182] > variable_user_name: [27879998182] > variable_domain_name: [196.99.88.77] > variable_record_stereo: [true] > variable_default_gateway: [verso] > variable_default_areacode: [87] > variable_transfer_fallback_extension: [operator] > variable_sip-force-expires: [180] > variable_toll_allow: [domestic,international] > variable_accountcode: [tbaaaa] > variable_user_context: [sprof1] > variable_effective_caller_id_name: [TickyBox 99999 Phone 99 Test] > variable_effective_caller_id_number: [879998182] > variable_outbound_caller_id_name: [879998150] > variable_outbound_caller_id_number: [879998150] > variable_sip_from_user: [27879998182] > variable_sip_from_uri: [27879998182 at 196.99.88.77] > variable_sip_from_host: [196.99.88.77] > variable_sip_from_user_stripped: [27879998182] > variable_sip_from_tag: [196.99.88.77] > variable_sofia_profile_name: [sprof1] > variable_sofia_profile_domain_name: [196.99.88.77] > variable_sip_req_user: [0879998154] > variable_sip_req_uri: [0879998154 at 196.99.88.77] > variable_sip_req_host: [196.99.88.77] > variable_sip_to_user: [0879998154] > variable_sip_to_uri: [0879998154 at 196.99.88.77] > variable_sip_to_host: [196.99.88.77] > variable_sip_contact_user: [27879998182] > variable_sip_contact_port: [22034] > variable_sip_contact_uri: [27879998182 at 172.16.0.63:22034] > variable_sip_contact_host: [172.16.0.63] > variable_channel_name: [sofia/sprof1/27879998182 at 196.99.88.77] > variable_sip_call_id: [xr125298731411533c30039109e1921f at 192.168.10.1] > variable_sip_user_agent: [BrokenPhone/1.4.2] > variable_sip_via_host: [172.16.0.63] > variable_sip_via_port: [22034] > variable_sip_via_rport: [36745] > variable_presence_id: [27879998182 at 196.99.88.77] > variable_switch_r_sdp: [v=0 > o=2787999818 2265 2267 IN IP4 172.16.0.63 > s=Broken > c=IN IP4 172.16.0.63 > t=0 0 > m=audio 29081 RTP/AVP 9 18 101 > a=rtpmap:9 G722/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a > = > candidates > :-1564465265,172.16.0.63:29081,192.168.10.1:29081,192.168.20.1:29081 > ] > variable_outboundcontext: [setupprepaycall] > variable_remote_media_ip: [172.16.0.63] > variable_remote_media_port: [29081] > variable_read_codec: [G722] > variable_read_rate: [16000] > variable_write_codec: [G722] > variable_write_rate: [16000] > variable_local_media_ip: [196.99.88.77] > variable_local_media_port: [16932] > variable_endpoint_disposition: [EARLY MEDIA] > variable_sip_nat_detected: [true] > variable_api_hangup_hook: [jsapi::completecall.js] > variable_talktime: [6870] > variable_action: [allow] > variable_status: [allowed] > variable_integer: [102] > variable_fraction: [85] > variable_saytalktime: [60:0] > variable_schedparms: [+3600 tbhangupwarn XML hangupwarn] > variable_bridgejscb: [{api_hangup_hook=jsapi::completecall.js}] > variable_max_forwards: [67] > variable_current_application: [info] > > > > 2009-03-16 14:37:59 [WARNING] mod_sofia.c:739 sofia_read_frame() We > were told to use ptime 20 but what they meant to say was 820 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sipura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who knows what will happen.. > 2009-03-16 14:37:59 [WARNING] switch_core_codec.c:499 > switch_core_codec_init() Codec G722 Exists but not at the desired > implementation. 8000hz 820ms > 2009-03-16 14:37:59 [ERR] sofia_glue.c:1700 > sofia_glue_tech_set_codec() Can't load codec? > 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 > switch_core_session_read_frame() sofia/ > sprof1/27879998182 at 196.99.88.77 has no read codec. > 2009-03-16 14:37:59 [ERR] switch_core_io.c:585 > switch_core_session_write_frame() sofia/sprof1/27879998182 at 196.99.88.77 > has no write codec. > 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 > switch_core_session_read_frame() sofia/ > sprof1/27879998182 at 196.99.88.77 has no read codec. > 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 > switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182 at 196.99.88.77 > to XML[8154 at toregext] > 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing MeMe->8154 in context toregext > 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 > execute_extension::dx XML features > 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/ > usr/local/freeswitch/recordings/27879998182.2009-03-16-14-37-59.wav > 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 > switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 > execute_extension::cf XML features > > > > 2009-03-16 14:38:00 [NOTICE] switch_channel.c:592 > switch_channel_set_name() New Channel sofia/voxwan/8154 > [48c2f400-1227-11de-8b8e-1789e43c417d] > Segmentation fault (core dumped) > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/de5c4fc0/attachment-0002.html From brian at freeswitch.org Mon Mar 16 06:48:03 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 08:48:03 -0500 Subject: [Freeswitch-users] Core Dump on receiving a call from device with 'broken' G.722 codec. In-Reply-To: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED73497A30@xena.internal.datapro.co.za> Message-ID: http://wiki.freeswitch.org/wiki/Reporting_Bugs Keith, Please read the link above... open a jira and collect a sip trace of this also and "attach" it. /b On Mar 16, 2009, at 8:36 AM, Keith Laaks wrote: > Hi, > > I am on fs 1.0.trunk (12530M) testing G.722 and found that when > using a ?broken? configuration from a softphone configured for G. > 722, I get the warning on the cli: > > ?We were told to use ptime 20 but what they meant to say was 820 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sipura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who knows what will happen.. > ? > when fs gets the invite, but then does a core dump when it tries to: > > > > > Below are some of the traces and info output from before the core > dump happens. > > I see this when I run gdb on the dumpfile. > > #0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, > bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/ > 8154%172.16.1.3", timelimit_sec=30, > table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, > caller_profile_override=0x0, ovars=0x0, flags=) > at src/switch_ivr_originate.c:1609 > 1609 if > (switch_core_codec_init(&write_codec, > > (gdb) frame 1 > #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so > > > I wonder if anybody else has seen this behavior? > > This happens when the destination phone is also G.722 capable > (policom). > If I change the ?frame per packet? setting in the softphone to 2 ? > All works OK (but the default is 1 ? so cant risk allowing G.722 if > it?s going to core dump fs if a user make a wrong configuration) > > > Best Regards > > Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/a2d6e21a/attachment-0002.html From fs at xenpad.eu Mon Mar 16 08:29:32 2009 From: fs at xenpad.eu (fs at xenpad.eu) Date: Mon, 16 Mar 2009 16:29:32 +0100 (CET) Subject: [Freeswitch-users] Problem with shortened local extensions In-Reply-To: References: Message-ID: Hi, I have a (probably dumb) question that I just spent over 5 hours on: I have a 1.0.2 version running with 6 extensions, 2 SIP trunks and gtalk OK. I tried to change the extensions from 100x to 10x (100-109 actually) I changed the info in the directory, hunted all references to patterns like 10[01][0-9], and replaced them with 10[0-9], and changed the DID routing to the new extension numbers. No errors at load time, sofia status profile internal shows all extensions registered; each extension can call it's own voicemail, and all provided examples (5000, 9998, ...) work OK. The catch is, it's not possible to call another extension, and the routing from incoming SIP trunks fail (the extension is not available, please oleave a message). I'm all out of ideas. I first tried that with a load of other changes; rolled back everything to the working setup, and just did this one change; all to no avail. What am I missing? TIA, Laurent From steve.d.ward at gmail.com Mon Mar 16 07:19:15 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 10:19:15 -0400 Subject: [Freeswitch-users] sip trunking question Message-ID: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> I'm trying to set up a sip trunk between a FS and * box, and right now I'm having trouble getting things set up so I make a call from a sip phone registered with my FS box to a sip phone registered w/ my Asterisk box. I have a gateway defined as in directory/default/example.com.xml and in my dialplan I'm trying to do a bridge w/ something like "sofia/gateway/${default_gateway}/12345." When I try to make the call I see from the console: ... New Channel sofia/external/12345 ... ... STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] ... Hangup sofia/external/12345 [CS_INIT] [DESTINATION_OUT_OF_ORDER] The Originate fails. I tried sticking to what the instructions laid out for this in the Connecting FS and Asterisk wiki page, so I'd appreciate some help in figuring out what's going on. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/6ba86a85/attachment-0002.html From kerrada2003 at yahoo.com Mon Mar 16 08:58:40 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 16 Mar 2009 08:58:40 -0700 (PDT) Subject: [Freeswitch-users] FS Database Message-ID: <135187.79162.qm@web33708.mail.mud.yahoo.com> Hi, Is it possible to access FS DB to retrieve data? Where can i find details about that? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/1ffeac71/attachment-0002.html From msc at freeswitch.org Mon Mar 16 08:59:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 08:59:18 -0700 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> Message-ID: <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> 2009/3/16 Steven Ward : > I'm trying to set up a sip trunk between a FS and * box, and right now I'm > having trouble getting things set up so I make a call from a sip phone > registered with my FS box to a sip phone registered w/ my Asterisk box. > > I have a gateway defined as in directory/default/example.com.xml and in my > dialplan I'm trying to do a bridge w/ something like > "sofia/gateway/${default_gateway}/12345." > > When I try to make the call I see from the console: > > ... New Channel sofia/external/12345 ... > ... STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] > ... Hangup sofia/external/12345 [CS_INIT] [DESTINATION_OUT_OF_ORDER] What is your network setup? The gateway you created is using the external profile and trying to do a STUN lookup. Is that what you are trying to do? Just confirming. -MC From msc at freeswitch.org Mon Mar 16 09:06:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 09:06:05 -0700 Subject: [Freeswitch-users] Problem with shortened local extensions In-Reply-To: References: Message-ID: <87f2f3b90903160906q693370c8s6b68eafd528f6401@mail.gmail.com> On Mon, Mar 16, 2009 at 8:29 AM, wrote: > ? Hi, > > ? I have a (probably dumb) question that I just spent over 5 hours on: > I have a 1.0.2 version running with 6 extensions, 2 SIP trunks and gtalk OK. Ouch! Any way you could update? We are on the verge of releasing 1.0.4; 1.0.2 is OLD. :) > > ? I tried to change the extensions from 100x to 10x (100-109 actually) > ? I changed the info in the directory, hunted all references to patterns > like 10[01][0-9], and replaced them with 10[0-9], and changed the DID > routing to the new extension numbers. > > ? No errors at load time, sofia status profile internal shows all > extensions registered; each extension can call it's own voicemail, and > all provided examples (5000, 9998, ...) work OK. > > ? The catch is, it's not possible to call another extension, and the > routing from incoming SIP trunks fail (the extension is not available, please > oleave a message). Best bet here is to read up on this page: http://wiki.freeswitch.org/wiki/Reporting_Bugs That will help you do stuff like this: turn on debugging (press F8 at the CLI) then make a test call, capture output, put it into a pastebin. I'm sure it's something basic, but without seeing what's happening it's hard to diagnose. Also, pastebin your default.xml dialplan file, one or more of your directory files, like 100.xml (or whatever you named them), and finally do "sofia status profile internal" at the CLI and pastebin the output. -MC > > ? I'm all out of ideas. I first tried that with a load of other changes; > rolled back everything to the working setup, and just did this one > change; all to no avail. > > ? What am I missing? > > ? TIA, > ? ?Laurent > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Mon Mar 16 09:24:34 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Mar 2009 12:24:34 -0400 Subject: [Freeswitch-users] FS Database In-Reply-To: <135187.79162.qm@web33708.mail.mud.yahoo.com> References: <135187.79162.qm@web33708.mail.mud.yahoo.com> Message-ID: <1BF9A176-1CE8-4D53-894B-E456DA8897A0@jerris.com> Which data? On Mar 16, 2009, at 11:58 AM, Ali Al-Rubaie wrote: > Hi, > > Is it possible to access FS DB to retrieve data? Where can i find > details about that? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d0d0a1b0/attachment-0002.html From msc at freeswitch.org Mon Mar 16 09:28:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 09:28:17 -0700 Subject: [Freeswitch-users] FS Database In-Reply-To: <135187.79162.qm@web33708.mail.mud.yahoo.com> References: <135187.79162.qm@web33708.mail.mud.yahoo.com> Message-ID: <87f2f3b90903160928u54583af6r48e1ba0a7f41b43e@mail.gmail.com> > Is it possible to access FS DB to retrieve data? Where can i find details > about that? Could you be a little more specific? Also, on a standard Linux/Unix install you could check here: /usr/local/freeswitch/db/*db -MC From fs at xenpad.eu Mon Mar 16 09:36:27 2009 From: fs at xenpad.eu (fs at xenpad.eu) Date: Mon, 16 Mar 2009 17:36:27 +0100 (CET) Subject: [Freeswitch-users] Problem with shortened local extensions In-Reply-To: <87f2f3b90903160906q693370c8s6b68eafd528f6401@mail.gmail.com> References: <87f2f3b90903160906q693370c8s6b68eafd528f6401@mail.gmail.com> Message-ID: Hi, On Mon, 16 Mar 2009, Michael Collins wrote: >> ? I have a (probably dumb) question that I just spent over 5 hours on: >> I have a 1.0.2 version running with 6 extensions, 2 SIP trunks and gtalk OK. > > Ouch! Any way you could update? We are on the verge of releasing > 1.0.4; 1.0.2 is OLD. :) Then I'll skip 1.0.3 and wait for 1.0.4 ;) > That will help you do stuff like this: > turn on debugging (press F8 at the CLI) then make a test call, capture > output, put it into a pastebin. OK -- this is a live system; I worked on it this weekend. I'll try again with the short extension and collect all data. Should 1.0.4 be out in the meantime, I'll upgrade before I try. Cheers, Laurent -- +-----------------------------------------------------------------------+ | E-mail Home page http://case.jorune.net/ | | We do what we must because we can. | +-----------------------------------------------------------------------+ From kerrada2003 at yahoo.com Mon Mar 16 09:46:04 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 16 Mar 2009 09:46:04 -0700 (PDT) Subject: [Freeswitch-users] FS DB In-Reply-To: Message-ID: <595500.67715.qm@web33706.mail.mud.yahoo.com> Actually, I'm trying to explore what data is stored in the DB and how can we retrieve it, if it is posible. You may write some scripts to retrieve the data and use it in other supporting application. Message: 7 Date: Mon, 16 Mar 2009 12:24:34 -0400 From: Michael Jerris Subject: Re: [Freeswitch-users] FS Database To: freeswitch-users at lists.freeswitch.org Message-ID: <1BF9A176-1CE8-4D53-894B-E456DA8897A0 at jerris.com> Content-Type: text/plain; charset="us-ascii" Which data? On Mar 16, 2009, at 11:58 AM, Ali Al-Rubaie wrote: > Hi, > > Is it possible to access FS DB to retrieve data? Where can i find > details about that? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d0d0a1b0/attachment.html ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 33, Issue 67 ************************************************ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/84be11c8/attachment-0002.html From Mark.Tabron at rnid-typetalk.org.uk Mon Mar 16 09:53:25 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Mon, 16 Mar 2009 16:53:25 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room. So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be. Any ideas? Pastebin debug output is in my reply below. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 13 March 2009 14:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! http://pastebin.freeswitch.org/7751 Will setup an IRC client and see if I can log onto the channel. Thanks again! -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 March 2009 16:50 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 > My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so > apologies if some of my terminology isn't quite correct. Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > > > > Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've > hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the > card is green and wanrouter is installed and up in TDM_API mode, with the > connection status showing as connected. ?Configured Openzap for 9 b and 1 d > channel as described in Freeswitch Wiki. Then created a diaplan to fire off > any calls preceded by 9 to the next available openzap channel. Looks good so far... > The problem I have is when I initiate an external call (using 9xxxxxxx) from > an extension I can see Freeswitch allocating the call to the next available > channel but then the just sits there and times out after 1 minute. With the > cause stated as ORIGINATOR_CANCEL (guessing this is the time out) okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has lots of useful information for how to gather log information, how to use the pastebin, etc. Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers: To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org). Welcome to the wonderful world of telephony debugging! -MC P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.) _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From steve.d.ward at gmail.com Mon Mar 16 10:15:25 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 13:15:25 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> Message-ID: <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> Yes, the obvious is the case. :) I don't want to do a STUN lookup - the two machines are on the same LAN. What's the best way to get the gateway to not do a STUN lookup? Do I need to disable STUN for the external profile or make this gateway use a different profile? Thanks. SW On Mon, Mar 16, 2009 at 11:59 AM, Michael Collins wrote: > 2009/3/16 Steven Ward : > > I'm trying to set up a sip trunk between a FS and * box, and right now > I'm > > having trouble getting things set up so I make a call from a sip phone > > registered with my FS box to a sip phone registered w/ my Asterisk box. > > > > I have a gateway defined as in directory/default/example.com.xml and in > my > > dialplan I'm trying to do a bridge w/ something like > > "sofia/gateway/${default_gateway}/12345." > > > > When I try to make the call I see from the console: > > > > ... New Channel sofia/external/12345 ... > > ... STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] > > ... Hangup sofia/external/12345 [CS_INIT] [DESTINATION_OUT_OF_ORDER] > > What is your network setup? The gateway you created is using the > external profile and trying to do a STUN lookup. Is that what you are > trying to do? Just confirming. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/f5936e5f/attachment-0002.html From msc at freeswitch.org Mon Mar 16 10:15:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 10:15:53 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> Any chance you can give one of us access to this system? Best thing to do would be to join #openzap on irc.freenode.net. -MC (IRC: mercutioviz) On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron wrote: > Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room. > > So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be. > > Any ideas? Pastebin debug output is in my reply below. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron > Sent: 13 March 2009 14:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). > > Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! > > http://pastebin.freeswitch.org/7751 > > Will setup an IRC client and see if I can log onto the channel. > > Thanks again! > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 12 March 2009 16:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > >> My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so >> apologies if some of my terminology isn't quite correct. > > Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > >> >> >> >> Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've >> hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the >> card is green and wanrouter is installed and up in TDM_API mode, with the >> connection status showing as connected. ?Configured Openzap for 9 b and 1 d >> channel as described in Freeswitch Wiki. Then created a diaplan to fire off >> any calls preceded by 9 to the next available openzap channel. > > Looks good so far... > >> The problem I have is when I initiate an external call (using 9xxxxxxx) from >> an extension I can see Freeswitch allocating the call to the next available >> channel but then the just sits there and times out after 1 minute. With the >> cause stated as ORIGINATOR_CANCEL (guessing this is the time out) > > okay, some debugging info will be useful. Please read this wiki page first: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It has lots of useful information for how to gather log information, > how to use the pastebin, etc. > > Specifically for this issue you'll need to use the pastebin because > there will be so much information. Here are some pointers: > > To see what's happening with openzap you'll need to use the "oz list" > and "oz dump 1" at the command line (CLI). You'll also need to turn on > debugging so that PRI messages show up. You'll need to capture the > output on the CLI and put it into the pastebin. > (http://pastebin.freeswitch.org). > > Welcome to the wonderful world of telephony debugging! > -MC > > P.S. - We have a few IRC channels where you can join to get more > real-time support: > #freeswitch and #openzap on irc.freenode.net. (More details are in the > wiki page I mentioned above.) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > -------------------------------------------------------------------------------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > -------------------------------------------------------------------------------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon Mar 16 10:24:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 10:24:15 -0700 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> Message-ID: <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> 2009/3/16 Steven Ward : > Yes, the obvious is the case.? :) I don't want to do a STUN lookup - the two > machines are on the same LAN. > > What's the best way to get the gateway to not do a STUN lookup?? Do I need > to disable STUN for the external > profile or make this gateway use a different profile? In which directory do you create your gateway file? If you created it in sip_profiles/external/ then try moving it over to sip_profiles/internal/ and see what happens... -MC P.S. - You could also disable STUN on your external profile, but since your two boxes are on the same LAN I would suggestion that the "proper" way to handle this situation is to have your gateway use the internal profile. From keithl at voxtelecom.co.za Mon Mar 16 10:25:06 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Mon, 16 Mar 2009 19:25:06 +0200 Subject: [Freeswitch-users] Module Mod Native File - How to handle multiple rates under the same codec. Message-ID: <1B99233662E2104983E3550185D3ED73497A49@xena.internal.datapro.co.za> Hi, To minimize/eliminate transcoding, I am using mod native file, with a set of transcoded prompts with the appropriate set of file extensions. Everything works as advertised when using the traditional codecs such as pcma, g729, gsm. But speex support is a bit of a challenge. Freeswitch supports: speex at 8000h@20i, speex at 16000h@20i, speex at 32000h@20i. When I setup calls using these various codec flavors, I can see using info (and of course detect by ear) that indeed the call is running at the different 'rates'. I see that regardless of which one I use, the "variable_read_codec" and "variable_write_codec" remains "SPEEX", but depending on the flavor, a "variable_read_rate" and "variable_write_rate" of either 8000,16000 or 32000. But when I try play a file when in 16000 or 32000, I get: 2009-03-15 16:31:13 [INFO] mod_native_file.c:81 native_file_file_open() Opening File [/usr/local/freeswitch/sounds/en/us/callie/all/16000/SUCCESS.SPEEX] 8000hz 2009-03-15 16:31:13 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match. I created my SPEEX files using: speexenc -w (note the -w option for 16kHz wideband) So, even though the call is setup using a wideband 16kHz codec, it appears that mod native file is expecting a 8kHz file for all the SPEEX flavors. What am I missing here? Is this module limited to 8KHz rates? I am on 1.0.trunk (12530M). I have not yet looked at these codecs: G7221 at 16000h, G7221 at 32000h, CELT at 32000h, CELT at 48000h, but as these also have multiple rates for the same codec - I expect same issue. I am using ${ variable_read_rate } in the filename path, so fs looks at a set of files encoded with a matching sample rate. But looks like it's always looking for a 8KHz file. If you have had any experience with this, please let me have your advice. Thanks Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/f4c71603/attachment-0002.html From mrene_lists at avgs.ca Mon Mar 16 10:26:32 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 16 Mar 2009 13:26:32 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> Message-ID: <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> The reason its using stun is because your external-sip-ip and external- rtp-ip params are starting with stun: As Michael says, the external profile is meant to do nat-traversal, if you dont need it, use the internal one. Math On 16-Mar-09, at 1:24 PM, Michael Collins wrote: > 2009/3/16 Steven Ward : >> Yes, the obvious is the case. :) I don't want to do a STUN lookup >> - the two >> machines are on the same LAN. >> >> What's the best way to get the gateway to not do a STUN lookup? Do >> I need >> to disable STUN for the external >> profile or make this gateway use a different profile? > > In which directory do you create your gateway file? If you created it > in sip_profiles/external/ then try moving it over to > sip_profiles/internal/ and see what happens... > > -MC > > P.S. - You could also disable STUN on your external profile, but since > your two boxes are on the same LAN I would suggestion that the > "proper" way to handle this situation is to have your gateway use the > internal profile. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steve.d.ward at gmail.com Mon Mar 16 10:27:52 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 13:27:52 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> Message-ID: <4ea6e8f20903161027q5d2c18bfmf85c493ac3f84cf9@mail.gmail.com> Thanks. I created the gateway file in conf/directory/default/ On Mon, Mar 16, 2009 at 1:24 PM, Michael Collins wrote: > 2009/3/16 Steven Ward : > > Yes, the obvious is the case. :) I don't want to do a STUN lookup - the > two > > machines are on the same LAN. > > > > What's the best way to get the gateway to not do a STUN lookup? Do I > need > > to disable STUN for the external > > profile or make this gateway use a different profile? > > In which directory do you create your gateway file? If you created it > in sip_profiles/external/ then try moving it over to > sip_profiles/internal/ and see what happens... > > -MC > > P.S. - You could also disable STUN on your external profile, but since > your two boxes are on the same LAN I would suggestion that the > "proper" way to handle this situation is to have your gateway use the > internal profile. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/480d5bd5/attachment-0002.html From steve.d.ward at gmail.com Mon Mar 16 10:39:32 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 13:39:32 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> Message-ID: <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> I simply moved the file defining the gateway to conf/sip_profiles/internal Well, when calling from extension 1000 to 70904, what I see on the console (debug mode) is: 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/1000 at pbx-sip-3.usa.internal.net Execute bridge(sofia/gateway/${default_gateway}/70904) 2009-03-16 13:35:39 [DEBUG] switch_core_session.c:1286 switch_core_session_exec() sofia/internal/1000 at pbx-sip-3.usa.internal.netExpanded String bridge(sofia/gateway/ pbx-sip-4.usa.internal.net/70904) 2009-03-16 13:35:39 [ERR] mod_sofia.c:2379 sofia_outgoing_channel() Invalid Gateway 2009-03-16 13:35:39 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2009-03-16 13:35:39 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2009-03-16 13:35:39 [DEBUG] switch_ivr_originate.c:2014 switch_ivr_originate() Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT] 2009-03-16 13:35:39 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: INVALID_NUMBER_FORMAT What else am I missing and not doing right? Thanks again for your help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/18d7c427/attachment-0002.html From brian at freeswitch.org Mon Mar 16 10:44:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 12:44:18 -0500 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> Message-ID: <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> I would almost bet your xml is wrong when you moved it.. care to share that little bit of info? /b On Mar 16, 2009, at 12:39 PM, Steven Ward wrote: > I simply moved the file defining the gateway to conf/sip_profiles/ > internal > > Well, when calling from extension 1000 to 70904, what I see on the > console (debug mode) is: > > 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1000 at pbx-sip-3.usa.internal.net > Execute bridge(sofia/gateway/${default_gateway}/70904) > 2009-03-16 13:35:39 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1000 at pbx-sip-3.usa.internal.net > Expanded String bridge(sofia/gateway/pbx-sip-4.usa.internal.net/ > 70904) > 2009-03-16 13:35:39 [ERR] mod_sofia.c:2379 sofia_outgoing_channel() > Invalid Gateway > 2009-03-16 13:35:39 [NOTICE] mod_sofia.c:2591 > sofia_outgoing_channel() Close Channel N/A [CS_NEW] > 2009-03-16 13:35:39 [ERR] switch_ivr_originate.c:1425 > switch_ivr_originate() Cannot create outgoing channel of type > [sofia] cause: [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [DEBUG] switch_ivr_originate.c:2014 > switch_ivr_originate() Originate Resulted in Error Cause: 28 > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [INFO] mod_dptools.c:1998 > audio_bridge_function() Originate Failed. Cause: > INVALID_NUMBER_FORMAT > > > What else am I missing and not doing right? Thanks again for your > help. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/4d982fb9/attachment-0002.html From msc at freeswitch.org Mon Mar 16 10:47:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Mar 2009 10:47:32 -0700 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> Message-ID: <87f2f3b90903161047n4dbcf8c4i6e4924e9d9729a09@mail.gmail.com> 2009/3/16 Steven Ward : > I simply moved the file defining the gateway to conf/sip_profiles/internal > > Well, when calling from extension 1000 to 70904, what I see on the console > (debug mode) is: > > 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() > sofia/internal/1000 at pbx-sip-3.usa.internal.net Execute > bridge(sofia/gateway/${default_gateway}/70904) > 2009-03-16 13:35:39 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1000 at pbx-sip-3.usa.internal.net > Expanded String bridge(sofia/gateway/pbx-sip-4.usa.internal.net/70904) > 2009-03-16 13:35:39 [ERR] mod_sofia.c:2379 sofia_outgoing_channel() Invalid > Gateway ^^^^^^^^^^^^^^^^ There's your key. Invalid gateway means just that: you're dialing a gw that doesn't exist. What is your gateway name in the file? Is it really "pbx-sip-4.usa.internal.net" ? does that resolve to an internal IP address? -MC > 2009-03-16 13:35:39 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() Close > Channel N/A [CS_NEW] > 2009-03-16 13:35:39 [ERR] switch_ivr_originate.c:1425 switch_ivr_originate() > Cannot create outgoing channel of type [sofia] cause: > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [DEBUG] switch_ivr_originate.c:2014 > switch_ivr_originate() Originate Resulted in Error Cause: 28 > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [INFO] mod_dptools.c:1998 audio_bridge_function() > Originate Failed.? Cause: INVALID_NUMBER_FORMAT > > > What else am I missing and not doing right?? Thanks again for your help. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From steve.d.ward at gmail.com Mon Mar 16 10:51:20 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 13:51:20 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> Message-ID: <4ea6e8f20903161051k1c6609ecp1b0ec57a68810e8@mail.gmail.com> Sure thing. Here it is: In vars.conf I supplied the variables' values: 2009/3/16 Brian West > I would almost bet your xml is wrong when you moved it.. care to share that > little bit of info? > /b > > On Mar 16, 2009, at 12:39 PM, Steven Ward wrote: > > I simply moved the file defining the gateway to > conf/sip_profiles/internal > > Well, when calling from extension 1000 to 70904, what I see on the console > (debug mode) is: > > 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() > sofia/internal/1000 at pbx-sip-3.usa.internal.net Execute > bridge(sofia/gateway/${default_gateway}/70904) > 2009-03-16 13:35:39 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1000 at pbx-sip-3.usa.internal.net Expanded > String bridge(sofia/gateway/pbx-sip-4.usa.internal.net/70904) > 2009-03-16 13:35:39 [ERR] mod_sofia.c:2379 sofia_outgoing_channel() Invalid > Gateway > 2009-03-16 13:35:39 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() > Close Channel N/A [CS_NEW] > 2009-03-16 13:35:39 [ERR] switch_ivr_originate.c:1425 > switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [DEBUG] switch_ivr_originate.c:2014 > switch_ivr_originate() Originate Resulted in Error Cause: 28 > [INVALID_NUMBER_FORMAT] > 2009-03-16 13:35:39 [INFO] mod_dptools.c:1998 audio_bridge_function() > Originate Failed. Cause: INVALID_NUMBER_FORMAT > > > What else am I missing and not doing right? Thanks again for your help. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/1bdd6466/attachment-0002.html From brian at freeswitch.org Mon Mar 16 10:53:32 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 12:53:32 -0500 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <4ea6e8f20903161051k1c6609ecp1b0ec57a68810e8@mail.gmail.com> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> <4ea6e8f20903161051k1c6609ecp1b0ec57a68810e8@mail.gmail.com> Message-ID: <44B6617A-ED52-44DF-82F1-B931BDD09F7F@freeswitch.org> First off since its not in the user directory anymore you'll have to unwrap the gateway from inside the user tags ;) /b On Mar 16, 2009, at 12:51 PM, Steven Ward wrote: > Sure thing. Here it is: > > > > > > > > > > > > > > > > > > > In vars.conf I supplied the variables' values: > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/250e0569/attachment-0002.html From steve.d.ward at gmail.com Mon Mar 16 11:08:19 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Mon, 16 Mar 2009 14:08:19 -0400 Subject: [Freeswitch-users] sip trunking question In-Reply-To: <44B6617A-ED52-44DF-82F1-B931BDD09F7F@freeswitch.org> References: <4ea6e8f20903160719x3b4e1c1evbdb0a26e51e1d40c@mail.gmail.com> <87f2f3b90903160859t4374ca9cocc79789b6ccb5468@mail.gmail.com> <4ea6e8f20903161015m140de023oed278140bd114c11@mail.gmail.com> <87f2f3b90903161024ha392147l8860b120c3d5487e@mail.gmail.com> <49EFBC37-ACCF-46DE-956F-1856DBC70EC8@avgs.ca> <4ea6e8f20903161039g3ba54efasce8fc68e0aa7070e@mail.gmail.com> <9C75E621-0EEC-41E2-84DA-C5031DD85E63@freeswitch.org> <4ea6e8f20903161051k1c6609ecp1b0ec57a68810e8@mail.gmail.com> <44B6617A-ED52-44DF-82F1-B931BDD09F7F@freeswitch.org> Message-ID: <4ea6e8f20903161108l2486ffbfl6f73acf6c298f7a6@mail.gmail.com> Heh heh. Guess it pays not to rush. :) Got it working now - without registering. But another thing - what if I want to set my two boxes up for registering? I see that I can set my register parameter to true, but how do I control the register string that's sent to the other box? 2009/3/16 Brian West > First off since its not in the user directory anymore you'll have to unwrap > the gateway from inside the user tags ;) > /b > > On Mar 16, 2009, at 12:51 PM, Steven Ward wrote: > > Sure thing. Here it is: > > > > > > > > value="$${default_provider_from_domain}"/> > > > > > > > > > > > In vars.conf I supplied the variables' values: > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/ac3f3aed/attachment-0002.html From kerrada2003 at yahoo.com Mon Mar 16 11:57:57 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 16 Mar 2009 11:57:57 -0700 (PDT) Subject: [Freeswitch-users] Outbound Codec Message-ID: <927033.2754.qm@web33705.mail.mud.yahoo.com> Hi, Is there a way to configure FS to offer a specific codec for B regardless of the codec chosen for? A leg? So, when FS invites B, it will offer only one codec specified by the administrator. Thanks, ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/4bcd12a7/attachment-0002.html From brian at freeswitch.org Mon Mar 16 12:06:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 14:06:18 -0500 Subject: [Freeswitch-users] Outbound Codec In-Reply-To: <927033.2754.qm@web33705.mail.mud.yahoo.com> References: <927033.2754.qm@web33705.mail.mud.yahoo.com> Message-ID: <3CF232FA-7502-4A4C-B037-1DB65D410A61@freeswitch.org> {absolute_codec_string=PCMU}sofia/blah/blah at blah /b On Mar 16, 2009, at 1:57 PM, Ali Al-Rubaie wrote: > Hi, > > Is there a way to configure FS to offer a specific codec for B > regardless of the codec chosen for A leg? So, when FS invites B, it > will offer only one codec specified by the administrator. > > Thanks, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/c1027f65/attachment-0002.html From sicfslist at gmail.com Mon Mar 16 12:07:01 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 16 Mar 2009 14:07:01 -0500 Subject: [Freeswitch-users] Outbound Codec In-Reply-To: <927033.2754.qm@web33705.mail.mud.yahoo.com> References: <927033.2754.qm@web33705.mail.mud.yahoo.com> Message-ID: <35b355e90903161207j4b136d68pbbc9d055ca3697fb@mail.gmail.com> Yes. You can set this in the sip profile settings ... and then when it's called in the bridge statement it will just work. For example you take a call on profile external (which allows multiple codecs) and then bridge it via profile internal (which only allows one codec). SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/48aefee2/attachment-0002.html From ludovic.fouquet at bewan.com Mon Mar 16 10:55:25 2009 From: ludovic.fouquet at bewan.com (ludovic) Date: Mon, 16 Mar 2009 18:55:25 +0100 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? Message-ID: <49BE928D.3090509@bewan.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/3fbfae78/attachment-0002.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: trace_fs.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/3fbfae78/attachment-0002.txt From brian at freeswitch.org Mon Mar 16 13:08:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 15:08:18 -0500 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: <49BE928D.3090509@bewan.com> References: <49BE928D.3090509@bewan.com> Message-ID: This would be one thing to look at your DNS name isn't resolving correctly.. you might consider using dynamic DNS and you can then set the them to "host:myhost.dyndns.org" /b On Mar 16, 2009, at 12:55 PM, ludovic wrote: > 2009-03-16 18:29:42 [DEBUG] sofia.c:206 sofia_event_callback() event > [nua_r_invite] status [503][DNS Error] session: sofia/external/ > 0123456789 > 2009-03-16 18:29:42 [DEBUG] sofia.c:206 sofia_event_callback() event > [nua_i_state] status [503][DNS Error] session: sofia/external/ > 0123456789 From chris at fowler.cc Mon Mar 16 13:19:18 2009 From: chris at fowler.cc (Chris Fowler) Date: Mon, 16 Mar 2009 13:19:18 -0700 Subject: [Freeswitch-users] Possible memory / cpu leak Message-ID: <1237234758.32766.1305717113@webmail.messagingengine.com> Hi, I?ve been seeing an issue where FreeSWITCH?s CPU and memory utilization climb over time; a restart of FS clears up the problem. See graphs for the past week. http://cfowl.postinbox.com/fs.jpg Observed on the Release Candidate, and then upgraded to the current trunk a couple of times. Currently running version ?FreeSWITCH Version 1.0.trunk (12604)?. This is seen both when FS is being used (~200 calls/day, and over the weekend when ~5 calls/day). How can I best debug this? Thanks, Chris. From brian at freeswitch.org Mon Mar 16 14:37:28 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 16:37:28 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237234758.32766.1305717113@webmail.messagingengine.com> References: <1237234758.32766.1305717113@webmail.messagingengine.com> Message-ID: <81B45675-0B08-45FA-B0BB-A23C834A438B@freeswitch.org> Can you update to SVN trunk as of now? /b On Mar 16, 2009, at 3:19 PM, Chris Fowler wrote: > Hi, > > I?ve been seeing an issue where FreeSWITCH?s CPU and memory > utilization > climb over time; a restart of FS clears up the problem. > > See graphs for the past week. http://cfowl.postinbox.com/fs.jpg > > Observed on the Release Candidate, and then upgraded to the current > trunk a couple of times. Currently running version ?FreeSWITCH > Version > 1.0.trunk (12604)?. > > This is seen both when FS is being used (~200 calls/day, and over the > weekend when ~5 calls/day). > > How can I best debug this? > > Thanks, Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d16a6b7d/attachment-0002.html From brian at freeswitch.org Mon Mar 16 14:39:03 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 16:39:03 -0500 Subject: [Freeswitch-users] Don't forget about FreeSWITCH-biz mailing list. Message-ID: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz If you want to do business related posts... ads... discuss services and such that don't belong on the users or dev lists please join the freeswitch-dev list. Thanks, Brian West From jaybinks at gmail.com Mon Mar 16 15:27:29 2009 From: jaybinks at gmail.com (jay binks) Date: Tue, 17 Mar 2009 08:27:29 +1000 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237234758.32766.1305717113@webmail.messagingengine.com> References: <1237234758.32766.1305717113@webmail.messagingengine.com> Message-ID: what happens in your dialplan ? is is possible that you execute a script on each call, thats not being exited ? Jay On Tue, Mar 17, 2009 at 6:19 AM, Chris Fowler wrote: > Hi, > > I?ve been seeing an issue where FreeSWITCH?s CPU and memory utilization > climb over time; a restart of FS clears up the problem. > > See graphs for the past week. http://cfowl.postinbox.com/fs.jpg > > Observed on the Release Candidate, and then upgraded to the current > trunk a couple of times. Currently running version ?FreeSWITCH Version > 1.0.trunk (12604)?. > > This is seen both when FS is being used (~200 calls/day, and over the > weekend when ~5 calls/day). > > How can I best debug this? > > Thanks, Chris. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/b321575f/attachment-0002.html From chris at fowler.cc Mon Mar 16 15:42:03 2009 From: chris at fowler.cc (Chris Fowler) Date: Mon, 16 Mar 2009 15:42:03 -0700 Subject: [Freeswitch-users] Possible memory / cpu leak Message-ID: <1237243323.1988.1305741229@webmail.messagingengine.com> >> Jay : what happens in your dialplan ? Nothing special; no external script execution just default pattern matching to route to extensions (per the stock config). >> Brian: Can you update to SVN trunk as of now? Yup, I will pull the trunk and report back in 24 hours. Chris. From brian at freeswitch.org Mon Mar 16 15:49:04 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 17:49:04 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237243323.1988.1305741229@webmail.messagingengine.com> References: <1237243323.1988.1305741229@webmail.messagingengine.com> Message-ID: Happen to use voicemail a lot? ivr.conf.xml? /b On Mar 16, 2009, at 5:42 PM, Chris Fowler wrote: >>> Jay : what happens in your dialplan ? > Nothing special; no external script execution just default pattern > matching to route to extensions (per the stock config). > >>> Brian: Can you update to SVN trunk as of now? > Yup, I will pull the trunk and report back in 24 hours. > > Chris. From anthony.minessale at gmail.com Mon Mar 16 15:51:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Mar 2009 17:51:50 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237243323.1988.1305741229@webmail.messagingengine.com> References: <1237243323.1988.1305741229@webmail.messagingengine.com> Message-ID: <191c3a030903161551i72789854m680978cd063701ba@mail.gmail.com> nothing special is a bit vague. clearly something you are doing makes a difference. Perhaps you can explain any custom extensions you have or what you are doing a little better? install valgrind and run it for a while valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg then send us vg.log On Mon, Mar 16, 2009 at 5:42 PM, Chris Fowler wrote: > >> Jay : what happens in your dialplan ? > Nothing special; no external script execution just default pattern > matching to route to extensions (per the stock config). > > >> Brian: Can you update to SVN trunk as of now? > Yup, I will pull the trunk and report back in 24 hours. > > Chris. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/607776d2/attachment-0002.html From gservat at gmail.com Mon Mar 16 16:07:45 2009 From: gservat at gmail.com (Gonzalo Servat) Date: Mon, 16 Mar 2009 20:07:45 -0300 Subject: [Freeswitch-users] Don't forget about FreeSWITCH-biz mailing list. In-Reply-To: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> References: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> Message-ID: On Mon, Mar 16, 2009 at 6:39 PM, Brian West wrote: > http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz > > If you want to do business related posts... ads... discuss services > and such that don't belong on the users or dev lists please join the > freeswitch-dev list. > BUG! s/freeswitch-dev/freeswitch-biz ;-) - Gonzalo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/e6ce7b87/attachment-0002.html From brian at freeswitch.org Mon Mar 16 16:39:10 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 18:39:10 -0500 Subject: [Freeswitch-users] Don't forget about FreeSWITCH-biz mailing list. In-Reply-To: References: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> Message-ID: <52853F7E-2F51-4916-9FE3-F72F76C515FF@freeswitch.org> Ah yes... my bad... Thats what I get for doing two more things at once... Had a thread in my brain blocking! :P /b On Mar 16, 2009, at 6:07 PM, Gonzalo Servat wrote: > On Mon, Mar 16, 2009 at 6:39 PM, Brian West > wrote: > http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz > > If you want to do business related posts... ads... discuss services > and such that don't belong on the users or dev lists please join the > freeswitch-dev list. > > BUG! s/freeswitch-dev/freeswitch-biz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/43b33f17/attachment-0002.html From dujinfang at gmail.com Mon Mar 16 19:01:09 2009 From: dujinfang at gmail.com (seven) Date: Tue, 17 Mar 2009 10:01:09 +0800 Subject: [Freeswitch-users] FS DB In-Reply-To: <595500.67715.qm@web33706.mail.mud.yahoo.com> References: <595500.67715.qm@web33706.mail.mud.yahoo.com> Message-ID: <7BA5277B-9A6A-42C6-9388-3898E619C51A@gmail.com> In default configuration, all DB is located in FS/db/. It's sqlite DB, so if you have sqlite installed # apt-get install sqlite3 $ cd /usr/local/freeswitch/db $ sqlite3 core.db SQLite version 3.4.2 Enter ".help" for instructions sqlite> .help .bail ON|OFF Stop after hitting an error. Default OFF .databases List names and files of attached databases .dump ?TABLE? ... Dump the database in an SQL text format .echo ON|OFF Turn command echo on or off .exit Exit this program .explain ON|OFF Turn output mode suitable for EXPLAIN on or off. .header(s) ON|OFF Turn display of headers on or off .help Show this message .import FILE TABLE Import data from FILE into TABLE .indices TABLE Show names of all indices on TABLE .load FILE ?ENTRY? Load an extension library .mode MODE ?TABLE? Set output mode where MODE is one of: csv Comma-separated values column Left-aligned columns. (See .width) html HTML
code insert SQL insert statements for TABLE line One value per line list Values delimited by .separator string tabs Tab-separated values tcl TCL list elements .nullvalue STRING Print STRING in place of NULL values .output FILENAME Send output to FILENAME .output stdout Send output to the screen .prompt MAIN CONTINUE Replace the standard prompts .quit Exit this program .read FILENAME Execute SQL in FILENAME .schema ?TABLE? Show the CREATE statements .separator STRING Change separator used by output mode and .import .show Show the current values for various settings .tables ?PATTERN? List names of tables matching a LIKE pattern .timeout MS Try opening locked tables for MS milliseconds .width NUM NUM ... Set column widths for "column" mode sqlite> .tables aliases calls channels complete interfaces tasks sqlite> .schema channels CREATE TABLE channels ( uuid VARCHAR(255), created VARCHAR(255), created_epoch INTEGER, name VARCHAR(255), state VARCHAR(255), cid_name VARCHAR(255), cid_num VARCHAR(255), ip_addr VARCHAR(255), dest VARCHAR(255), application VARCHAR(255), application_data VARCHAR(255), dialplan VARCHAR(255), context VARCHAR(255), read_codec VARCHAR(255), read_rate VARCHAR(255), write_codec VARCHAR(255), write_rate VARCHAR(255) ); CREATE INDEX uuindex on channels (uuid); sqlite> select * from channels ...> ; sqlite> On Mar 17, 2009, at 12:46 AM, Ali Al-Rubaie wrote: > > Actually, I'm trying to explore what data is stored in the DB and > how can we retrieve it, if it is posible. You may write some scripts > to retrieve the data and use it in other supporting application. > > > Message: 7 > Date: Mon, 16 Mar 2009 12:24:34 -0400 > From: Michael Jerris > Subject: Re: [Freeswitch-users] FS Database > To: freeswitch-users at lists.freeswitch.org > Message-ID: <1BF9A176-1CE8-4D53-894B-E456DA8897A0 at jerris.com> > Content-Type: text/plain; charset="us-ascii" > > Which data? > > On Mar 16, 2009, at 11:58 AM, Ali Al-Rubaie wrote: > > > Hi, > > > > Is it possible to access FS DB to retrieve data? Where can i find > > details about that? > -------------- next part > -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090316/d0d0a1b0/attachment.html > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 33, Issue 67 > ************************************************ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/a7cc7e13/attachment-0002.html From dujinfang at gmail.com Mon Mar 16 19:06:23 2009 From: dujinfang at gmail.com (seven) Date: Tue, 17 Mar 2009 10:06:23 +0800 Subject: [Freeswitch-users] Don't forget about FreeSWITCH-biz mailing list. In-Reply-To: References: <69A33523-28B7-427A-8901-EDE8ED866EA6@freeswitch.org> Message-ID: <118441D4-BC68-4969-8BA8-AB2076E9F946@gmail.com> It's time to open a jira :D On Mar 17, 2009, at 7:07 AM, Gonzalo Servat wrote: > On Mon, Mar 16, 2009 at 6:39 PM, Brian West > wrote: > http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz > > If you want to do business related posts... ads... discuss services > and such that don't belong on the users or dev lists please join the > freeswitch-dev list. > > BUG! s/freeswitch-dev/freeswitch-biz > > ;-) > > - Gonzalo > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/c250d52c/attachment-0002.html From chris at fowler.cc Mon Mar 16 21:01:10 2009 From: chris at fowler.cc (Chris Fowler) Date: Mon, 16 Mar 2009 21:01:10 -0700 Subject: [Freeswitch-users] Possible memory / cpu leak Message-ID: <1237262470.17561.1305780929@webmail.messagingengine.com> >> Brian: Can you update to SVN trunk as of now? I updated - version reports: FreeSWITCH Version 1.0.trunk (12631) Only difference I note with this build is that upon "shutdown" FS now SegFaults. The mem/cpu usage continues to slowly climb. 2009-03-16 20:59:32 [CONSOLE] switch_loadable_module.c:1237 do_shutdown() Stopping: mod_spidermonkey Segmentation fault (core dumped) >> Anthony - nothing special is a bit vague. I modified the dial plan to accept extension in the 1000-1029 range Added DialByLast name using directory.lua (from the wiki) Modified the ivr config to play company specific greetings Voicemail is used a few times per day freeswitch at ip-10-250-155-18> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= internal profile sip:mod_sofia at xxx.xxx.xxx.xx:5060 RUNNING (0) external profile sip:mod_sofia at xxx.xxx.xxx.xx:5080 RUNNING (0) sip.flowroute.com gateway sip:xxxxxx at sip.flowroute.com REGED inphonex gateway sip:xxxxx at sip.inphonex.com REGED callwithus-did-xxxxxxxxxx gateway sip:xxxxxxx at east.callwithus.com REGED callwithus-did-xxxxxxxxxx gateway sip:xxxxxxx at east.callwithus.com REGED callwithus-did-xxxxxxxxxx gateway sip:xxxxxxx at east.callwithus.com REGED default alias internal ALIASED nat alias external ALIASED xxxxx.bbbb.aaa alias internal ALIASED outbound alias external ALIASED ================================================================================================= >> Anthony - valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg Nothing got logged, here's the output. Did I invoke valgrind incorrectly? ==32545== Memcheck, a memory error detector. ==32545== Copyright (C) 2002-2006, and GNU GPL'd, by Julian Seward et al. ==32545== Using LibVEX rev 1658, a library for dynamic binary translation. ==32545== Copyright (C) 2004-2006, and GNU GPL'd, by OpenWorks LLP. ==32545== Using valgrind-3.2.1, a dynamic binary instrumentation framework. ==32545== Copyright (C) 2000-2006, and GNU GPL'd, by Julian Seward et al. ==32545== For more details, rerun with: -v ==32545== ==32545== My PID = 32545, parent PID = 32511. Prog and args are: ==32545== /usr/local/freeswitch/bin/freeswitch ==32545== -vg ==32545== From brian at freeswitch.org Mon Mar 16 21:09:49 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Mar 2009 23:09:49 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237262470.17561.1305780929@webmail.messagingengine.com> References: <1237262470.17561.1305780929@webmail.messagingengine.com> Message-ID: Tip: ulimit -s 240, or freeswitch fork and execv out from under valgrind. /b On Mar 16, 2009, at 11:01 PM, Chris Fowler wrote: > > >>> Anthony - valgrind --tool=memcheck --log-file=vg.log --leak- >>> check=full --leak-resolution=high --show-reachable=yes /path/to/ >>> freeswitch -vg > > > Nothing got logged, here's the output. Did I invoke valgrind > incorrectly? > > ==32545== Memcheck, a memory error detector. > ==32545== Copyright (C) 2002-2006, and GNU GPL'd, by Julian Seward et > al. > ==32545== Using LibVEX rev 1658, a library for dynamic binary > translation. > ==32545== Copyright (C) 2004-2006, and GNU GPL'd, by OpenWorks LLP. > ==32545== Using valgrind-3.2.1, a dynamic binary instrumentation > framework. > ==32545== Copyright (C) 2000-2006, and GNU GPL'd, by Julian Seward et > al. > ==32545== For more details, rerun with: -v > ==32545== > ==32545== My PID = 32545, parent PID = 32511. Prog and args are: > ==32545== /usr/local/freeswitch/bin/freeswitch > ==32545== -vg > ==32545== From tntknight at gmail.com Mon Mar 16 21:55:35 2009 From: tntknight at gmail.com (Anthony Knight) Date: Tue, 17 Mar 2009 00:55:35 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards Message-ID: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> I'm thinking about doing a project that would use FreeSWITCH as an IVR, with callers being routed in by both ISDN PRI, and also SIP trunks, with occasional bridge calls between callers. I'm wondering in what use cases hardware echo cancellation on the PRI cards is needed. And does hardware echo cancellation work with OpenZap/FreeSWITCH? It looks like all the major cards (Sangoma, Digium, etc..) use Octasic Echo cancellation add-on cards. Is there any difference between brands? Any recommendations on PRI boards and whether I need to pay for echo cancellation are appreciated Thanks. Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/783854d7/attachment-0002.html From krice at freeswitch.org Mon Mar 16 22:35:53 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 17 Mar 2009 00:35:53 -0500 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> Message-ID: The cards that feature Hardware Echo Can?s work on hardware/driver level and are supported... From: Anthony Knight Reply-To: Date: Tue, 17 Mar 2009 00:55:35 -0400 To: Subject: [Freeswitch-users] echo cancellation on PRI cards I'm thinking about doing a project that would use?FreeSWITCH?as an IVR, with callers being routed in by both ISDN PRI, and also SIP trunks, with occasional bridge calls between callers. I'm wondering in what use cases hardware echo cancellation on the PRI cards is needed. ?And does hardware echo cancellation work with OpenZap/FreeSWITCH?? It looks like all the major cards (Sangoma, Digium, etc..) use Octasic Echo cancellation add-on cards. ?Is there any difference between brands? Any recommendations on PRI boards and whether I need to pay for echo cancellation are appreciated Thanks. Tony _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/5f7e30eb/attachment-0002.html From wasim at convergence.pk Mon Mar 16 22:48:20 2009 From: wasim at convergence.pk (Wasim Baig) Date: Tue, 17 Mar 2009 10:48:20 +0500 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> Message-ID: 2009/3/17 Anthony Knight I'm thinking about doing a project that would use FreeSWITCH as an IVR, with > callers being routed in by both ISDN PRI, and also SIP trunks, with > occasional bridge calls between callers. > > I'm wondering in what use cases hardware echo cancellation on the PRI cards > is needed. > When there is Echo being generated from the far end, usually in a bridged call. If you application is just an IVR, with no far end connectivity, then you shouldn't need an echo can. If you are bridging calls, then at some point you may need it, depending on what else is in the loop. > And does hardware echo cancellation work with OpenZap/FreeSWITCH? > Yes, it really has nothing to do with the software then, its handled by the card and its hardware driver. In Sangoma's case, by Wanpipe. > It looks like all the major cards (Sangoma, Digium, etc..) use Octasic Echo > cancellation add-on cards. Is there any difference between brands? > Sangoma has 1024 tap Octasic Echo Cans. Very nice they are indeed. > Any recommendations on PRI boards and whether I need to pay for echo > cancellation are appreciated > Unashamedly, Sangoma's. 100% of the cases where our customers have used Sangoma A10Xd vs A10X, they've been much happier with the quality on the line. Its a tad bit more $, but well worth it (especially in places with bad copper). -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/96550390/attachment-0002.html From kawarod at laposte.net Tue Mar 17 00:06:29 2009 From: kawarod at laposte.net (rod) Date: Tue, 17 Mar 2009 11:06:29 +0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BA844C.3010409@freeswitch.org> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> <49BA6608.70208@gmail.com> <49BA844C.3010409@freeswitch.org> Message-ID: <49BF4BF5.4080608@laposte.net> Hi, not too hard :p but it's just a bad habit when I write in my native language (french). I guess that this spelling is not too common for english speaker. I'll do my best next time to write it correctly. @tamas you are right, we could use limit_hash the same way as limit when not specifying the /rate @Mathieu did you suggest limit_hash is more scalable than limit? But I don't understand why limit_hash is not suitable for data replication (DB lookup for limit and memory for limit_hash??), even if I don't know how to do it with limit. regards. Raymond Chandler wrote: > Tamas wrote: > >> My guess is: pbm = problem :) >> >> > sure, but is it really that hard to spell all the way out? > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From kawarod at laposte.net Tue Mar 17 00:11:26 2009 From: kawarod at laposte.net (rod) Date: Tue, 17 Mar 2009 11:11:26 +0400 Subject: [Freeswitch-users] sip redirect contact variable no more available in SVN 12638 Message-ID: <49BF4D1E.9050108@laposte.net> Hi, running SVN r12638, I don't have access anymore to these 2 variables after a SIP 302 message, using info application: variable_sip_redirect_contact_user_0 variable_sip_redirect_contact_host_0 It was okay with SVN r12611. regards, rod From codecomplete at free.fr Tue Mar 17 02:02:29 2009 From: codecomplete at free.fr (Gilles) Date: Tue, 17 Mar 2009 10:02:29 +0100 Subject: [Freeswitch-users] Windows-compatible FXO PCI card? Message-ID: <7.0.1.0.2.20090317100224.02475630@free.fr> Hello For SOHO users, getting a second, Linux-based computer just to run a small voice server is overkill, so I'm thinking of selling an application based on the Windows version of Freeswitch. Instead of the Sangoma USB connector, I'd really prefer to sell them a PCI card, because it's less messy, and there's less chance of them disconnecting the hardware. I don't know of FXO PCI cards to connect an XP/Vista host to a phone line. Does someone know of such a thing? Thank you. From wasim at convergence.pk Tue Mar 17 02:21:42 2009 From: wasim at convergence.pk (Wasim Baig) Date: Tue, 17 Mar 2009 14:21:42 +0500 Subject: [Freeswitch-users] Windows-compatible FXO PCI card? In-Reply-To: <7.0.1.0.2.20090317100224.02475630@free.fr> References: <7.0.1.0.2.20090317100224.02475630@free.fr> Message-ID: On Tue, Mar 17, 2009 at 2:02 PM, Gilles wrote: I don't know of FXO PCI cards to connect an XP/Vista host to a phone > line. Does someone know of such a thing? Sangoma makes a low cost 4FXO, 1FXS. http://sangoma.com/products_and_solutions/hardware/analog_telephony/b600.html -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/273b5c79/attachment-0002.html From thomas.mangin at exa-networks.co.uk Tue Mar 17 02:27:13 2009 From: thomas.mangin at exa-networks.co.uk (Thomas Mangin) Date: Tue, 17 Mar 2009 09:27:13 +0000 Subject: [Freeswitch-users] Freeswitch and Kamailio (OpenSer) Integration In-Reply-To: <439e75680903060051y5021b292l76286dfe927d0337@mail.gmail.com> References: <439e75680903060051y5021b292l76286dfe927d0337@mail.gmail.com> Message-ID: <17216626-2A95-4BB1-8AD6-E7CD2C2B402C@exa-networks.co.uk> Yes it is possible but there is no documentation on how to do it. You will have to learn SIP and understand what you are doing. Forwarding the call to FS for nat may cause issue as FS will then not have direct connection to the phone and may not be able to always detect it is behind NAT. Have a look at SIP PATH Extension as it is what you need to force traffic back to KAM from FS. Regards, Thomas On 6 Mar 2009, at 08:51, Ramu wrote: > Hi All, > > I would like to setup freswitch and kamailio as follows: > > Kamailio acts as Proxy and Registrator > Freeswitch acts as a SBC and MediaServer (voicemail) > > Users will be reigstered to Kamailio > Kamailio forwards calls to FS to NAT > FS sends back INVITE to Kamailio > Kamailio will dial-out user. > > Bob calls Alice > Bob ==INVITE ==> Kamailio ==INVITE==> FS ==INVITE==> Kamailio > ==INVITE ==> Alice > > How can I achieve this scenario? Can you please direct me to any > documentation which is available? > > Thanks, > Ramu > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/8b68d0b7/attachment-0002.html From Mark.Tabron at rnid-typetalk.org.uk Tue Mar 17 02:31:01 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Tue, 17 Mar 2009 09:31:01 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though. I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it! Mark. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 16 March 2009 17:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Any chance you can give one of us access to this system? Best thing to do would be to join #openzap on irc.freenode.net. -MC (IRC: mercutioviz) On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron wrote: > Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room. > > So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be. > > Any ideas? Pastebin debug output is in my reply below. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron > Sent: 13 March 2009 14:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). > > Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! > > http://pastebin.freeswitch.org/7751 > > Will setup an IRC client and see if I can log onto the channel. > > Thanks again! > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 12 March 2009 16:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > >> My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so >> apologies if some of my terminology isn't quite correct. > > Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > >> >> >> >> Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've >> hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the >> card is green and wanrouter is installed and up in TDM_API mode, with the >> connection status showing as connected. ?Configured Openzap for 9 b and 1 d >> channel as described in Freeswitch Wiki. Then created a diaplan to fire off >> any calls preceded by 9 to the next available openzap channel. > > Looks good so far... > >> The problem I have is when I initiate an external call (using 9xxxxxxx) from >> an extension I can see Freeswitch allocating the call to the next available >> channel but then the just sits there and times out after 1 minute. With the >> cause stated as ORIGINATOR_CANCEL (guessing this is the time out) > > okay, some debugging info will be useful. Please read this wiki page first: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It has lots of useful information for how to gather log information, > how to use the pastebin, etc. > > Specifically for this issue you'll need to use the pastebin because > there will be so much information. Here are some pointers: > > To see what's happening with openzap you'll need to use the "oz list" > and "oz dump 1" at the command line (CLI). You'll also need to turn on > debugging so that PRI messages show up. You'll need to capture the > output on the CLI and put it into the pastebin. > (http://pastebin.freeswitch.org). > > Welcome to the wonderful world of telephony debugging! > -MC > > P.S. - We have a few IRC channels where you can join to get more > real-time support: > #freeswitch and #openzap on irc.freenode.net. (More details are in the > wiki page I mentioned above.) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > -------------------------------------------------------------------------------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > -------------------------------------------------------------------------------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mrene_lists at avgs.ca Tue Mar 17 03:10:57 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Mar 2009 06:10:57 -0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <49BF4BF5.4080608@laposte.net> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> <49BA6608.70208@gmail.com> <49BA844C.3010409@freeswitch.org> <49BF4BF5.4080608@laposte.net> Message-ID: <85B32876-7E49-4DDF-B92F-353DA9599DE9@avgs.ca> limit_hash uses a faster data structure then limit but works the same way for tne end-user. viens sur IRC si t'as des questions en francais =) Math On 17-Mar-09, at 3:06 AM, rod wrote: > Hi, > > not too hard :p > but it's just a bad habit when I write in my native language > (french). I > guess that this spelling is not too common for english speaker. > > I'll do my best next time to write it correctly. > > @tamas > you are right, we could use limit_hash the same way as limit when not > specifying the /rate > > @Mathieu > did you suggest limit_hash is more scalable than limit? But I don't > understand why limit_hash is not suitable for data replication (DB > lookup for limit and memory for limit_hash??), even if I don't know > how > to do it with limit. > > regards. > > Raymond Chandler wrote: >> Tamas wrote: >> >>> My guess is: pbm = problem :) >>> >>> >> sure, but is it really that hard to spell all the way out? >> >> -Ray >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ludovic.fouquet at bewan.com Tue Mar 17 04:22:13 2009 From: ludovic.fouquet at bewan.com (ludovic) Date: Tue, 17 Mar 2009 12:22:13 +0100 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: References: <49BE928D.3090509@bewan.com> Message-ID: <49BF87E5.5090809@bewan.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/0a4d5052/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: bewan100.jpg Type: image/jpeg Size: 3963 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/0a4d5052/attachment-0002.jpg From Mark.Tabron at rnid-typetalk.org.uk Tue Mar 17 04:24:31 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Tue, 17 Mar 2009 11:24:31 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 17 March 2009 09:31 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though. I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it! Mark. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 16 March 2009 17:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Any chance you can give one of us access to this system? Best thing to do would be to join #openzap on irc.freenode.net. -MC (IRC: mercutioviz) On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron wrote: > Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room. > > So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be. > > Any ideas? Pastebin debug output is in my reply below. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron > Sent: 13 March 2009 14:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). > > Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! > > http://pastebin.freeswitch.org/7751 > > Will setup an IRC client and see if I can log onto the channel. > > Thanks again! > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 12 March 2009 16:50 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > >> My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so >> apologies if some of my terminology isn't quite correct. > > Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE? > >> >> >> >> Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've >> hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the >> card is green and wanrouter is installed and up in TDM_API mode, with the >> connection status showing as connected. ?Configured Openzap for 9 b and 1 d >> channel as described in Freeswitch Wiki. Then created a diaplan to fire off >> any calls preceded by 9 to the next available openzap channel. > > Looks good so far... > >> The problem I have is when I initiate an external call (using 9xxxxxxx) from >> an extension I can see Freeswitch allocating the call to the next available >> channel but then the just sits there and times out after 1 minute. With the >> cause stated as ORIGINATOR_CANCEL (guessing this is the time out) > > okay, some debugging info will be useful. Please read this wiki page first: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It has lots of useful information for how to gather log information, > how to use the pastebin, etc. > > Specifically for this issue you'll need to use the pastebin because > there will be so much information. Here are some pointers: > > To see what's happening with openzap you'll need to use the "oz list" > and "oz dump 1" at the command line (CLI). You'll also need to turn on > debugging so that PRI messages show up. You'll need to capture the > output on the CLI and put it into the pastebin. > (http://pastebin.freeswitch.org). > > Welcome to the wonderful world of telephony debugging! > -MC > > P.S. - We have a few IRC channels where you can join to get more > real-time support: > #freeswitch and #openzap on irc.freenode.net. (More details are in the > wiki page I mentioned above.) > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > -------------------------------------------------------------------------------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > -------------------------------------------------------------------------------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From Prometheus001 at gmx.net Tue Mar 17 04:27:25 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 17 Mar 2009 12:27:25 +0100 Subject: [Freeswitch-users] Problem with shortened local extensions In-Reply-To: References: <87f2f3b90903160906q693370c8s6b68eafd528f6401@mail.gmail.com> Message-ID: <49BF891D.1090808@gmx.net> I also had problems with not reaching local extensions some time. I solved it by adding: to /usr/local/freeswitch/conf/directory/default.xml Best regards Peter fs at xenpad.eu schrieb: > Hi, > > On Mon, 16 Mar 2009, Michael Collins wrote: > >>> I have a (probably dumb) question that I just spent over 5 hours on: >>> I have a 1.0.2 version running with 6 extensions, 2 SIP trunks and >>> gtalk OK. >> >> Ouch! Any way you could update? We are on the verge of releasing >> 1.0.4; 1.0.2 is OLD. :) > > Then I'll skip 1.0.3 and wait for 1.0.4 ;) > >> That will help you do stuff like this: >> turn on debugging (press F8 at the CLI) then make a test call, capture >> output, put it into a pastebin. > OK -- this is a live system; I worked on it this weekend. > I'll try again with the short extension and collect all data. > Should 1.0.4 be out in the meantime, I'll upgrade before I try. > > Cheers, > Laurent > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveu at coppice.org Tue Mar 17 05:14:34 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 17 Mar 2009 20:14:34 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> Message-ID: <49BF942A.3030305@coppice.org> Wasim Baig wrote: > 2009/3/17 Anthony Knight > > > I'm thinking about doing a project that would use FreeSWITCH as an > IVR, with callers being routed in by both ISDN PRI, and also SIP > trunks, with occasional bridge calls between callers. > > I'm wondering in what use cases hardware echo cancellation on the > PRI cards is needed. > > > When there is Echo being generated from the far end, usually in a > bridged call. If you application is just an IVR, with no far end > connectivity, then you shouldn't need an echo can. If you are bridging > calls, then at some point you may need it, depending on what else is > in the loop. This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it they give very poor reliability detecting DTMF while the prompts are playing. If the system uses voice recognition, its reliability will be even worse. > And does hardware echo cancellation work with OpenZap/FreeSWITCH? > > > Yes, it really has nothing to do with the software then, its handled > by the card and its hardware driver. In Sangoma's case, by Wanpipe. > > > It looks like all the major cards (Sangoma, Digium, etc..) use > Octasic Echo cancellation add-on cards. Is there any difference > between brands? > > > Sangoma has 1024 tap Octasic Echo Cans. Very nice they are indeed. > > > Any recommendations on PRI boards and whether I need to pay for > echo cancellation are appreciated > > > Unashamedly, Sangoma's. 100% of the cases where our customers have > used Sangoma A10Xd vs A10X, they've been much happier with the quality > on the line. Its a tad bit more $, but well worth it (especially in > places with bad copper). If you use Sangoma make sure everything is up to date. People have had a lot of DTMF detection trouble with some revisions of the driver, or on board firmware, or possibly both. Clearly DTMF trouble would be pretty bad for an IVR. I didn't manage to trace which were the offending versions, but the current stuff is apparently OK. Steve From brian at freeswitch.org Tue Mar 17 06:53:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 08:53:07 -0500 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: <49BF87E5.5090809@bewan.com> References: <49BE928D.3090509@bewan.com> <49BF87E5.5090809@bewan.com> Message-ID: What I provided you was an example. I don't think you understood what I was talking about. In the settings for ext-sip-ip and ext-rtp-ip you'll have to use something like "host:yourdyndnshostname.blah.tld" Then set the sip-ip and rtp-ip to what ever is auto detected. /b On Mar 17, 2009, at 6:22 AM, ludovic wrote: > Thanks. > It seems that it comes from my sip provider. > when using my_host as my hostname, reg fails > when using my_host.com as my hostname, reg succeeds (my_host.com > does not exist as a domain internet) > when using ip address, reg succeeds. > > Tested with version 1.0.3 > > Is it a way to force the IP address to be used in SIP header instead > of hostname ? > > Thanks > > Ludovic From kzimnicki at gcdf.pl Tue Mar 17 02:17:13 2009 From: kzimnicki at gcdf.pl (Krzysztof Zimnicki) Date: Tue, 17 Mar 2009 10:17:13 +0100 Subject: [Freeswitch-users] [OpenZap] problem with TE220P Message-ID: <49BF6A99.6070308@gcdf.pl> Hi, I have problem with Digium TE220P. Everything works, i can call & talk, but everytime i have CRIT message: 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 When FS start show me ERR message: 2009-03-14 17:44:06 [ERR] Span:0 Q.921() Received UA frame in invalid state This is my config: cat zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span = 2,0,0,ccs,hdb3,crc4 bchan = 32-46,48-62 dchan = 47 loadzone = ru defaultzone=ru cat openzap.conf [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 cat openzap.conf.xml and FS start LOGS: 2009-03-14 17:44:06 [NOTICE] zap_io.c:2612 zap_global_init() Modules configured: 1 2009-03-14 17:44:06 [NOTICE] ozmod_zt.c:922 zt_init() Using Zaptel control device 2009-03-14 17:44:06 [INFO] zap_io.c:2433 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_zt.so [zt] 2009-03-14 17:44:06 [INFO] zap_io.c:2233 load_config() auto-loaded 'zt' 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:38 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 2 as OpenZAP device 1:2 fd:39 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 3 as OpenZAP device 1:3 fd:40 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 4 as OpenZAP device 1:4 fd:41 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 5 as OpenZAP device 1:5 fd:42 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 6 as OpenZAP device 1:6 fd:43 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 7 as OpenZAP device 1:7 fd:44 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 8 as OpenZAP device 1:8 fd:45 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 9 as OpenZAP device 1:9 fd:46 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 10 as OpenZAP device 1:10 fd:47 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 11 as OpenZAP device 1:11 fd:48 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 12 as OpenZAP device 1:12 fd:49 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 13 as OpenZAP device 1:13 fd:50 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 14 as OpenZAP device 1:14 fd:51 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 15 as OpenZAP device 1:15 fd:52 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 16 as OpenZAP device 1:16 fd:53 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 17 as OpenZAP device 1:17 fd:54 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 18 as OpenZAP device 1:18 fd:55 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 19 as OpenZAP device 1:19 fd:56 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 20 as OpenZAP device 1:20 fd:57 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 21 as OpenZAP device 1:21 fd:58 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 22 as OpenZAP device 1:22 fd:59 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 23 as OpenZAP device 1:23 fd:60 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 24 as OpenZAP device 1:24 fd:61 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 25 as OpenZAP device 1:25 fd:62 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 26 as OpenZAP device 1:26 fd:63 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 27 as OpenZAP device 1:27 fd:64 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 28 as OpenZAP device 1:28 fd:65 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 29 as OpenZAP device 1:29 fd:66 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 30 as OpenZAP device 1:30 fd:67 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 31 as OpenZAP device 1:31 fd:68 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 32 as OpenZAP device 2:1 fd:69 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 33 as OpenZAP device 2:2 fd:70 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 34 as OpenZAP device 2:3 fd:71 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 35 as OpenZAP device 2:4 fd:72 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 36 as OpenZAP device 2:5 fd:73 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 37 as OpenZAP device 2:6 fd:74 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 38 as OpenZAP device 2:7 fd:75 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 39 as OpenZAP device 2:8 fd:76 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 40 as OpenZAP device 2:9 fd:77 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 41 as OpenZAP device 2:10 fd:78 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 42 as OpenZAP device 2:11 fd:79 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 43 as OpenZAP device 2:12 fd:80 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 44 as OpenZAP device 2:13 fd:81 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 45 as OpenZAP device 2:14 fd:82 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 46 as OpenZAP device 2:15 fd:83 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 47 as OpenZAP device 2:16 fd:84 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 48 as OpenZAP device 2:17 fd:85 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 49 as OpenZAP device 2:18 fd:86 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 50 as OpenZAP device 2:19 fd:87 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 51 as OpenZAP device 2:20 fd:88 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 52 as OpenZAP device 2:21 fd:89 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 53 as OpenZAP device 2:22 fd:90 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 54 as OpenZAP device 2:23 fd:91 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 55 as OpenZAP device 2:24 fd:92 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 56 as OpenZAP device 2:25 fd:93 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 57 as OpenZAP device 2:26 fd:94 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 58 as OpenZAP device 2:27 fd:95 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 59 as OpenZAP device 2:28 fd:96 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 60 as OpenZAP device 2:29 fd:97 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 61 as OpenZAP device 2:30 fd:98 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 62 as OpenZAP device 2:31 fd:99 2009-03-14 17:44:06 [INFO] zap_io.c:2356 load_config() Configured 62 channel(s) 2009-03-14 17:44:06 [INFO] zap_io.c:2450 zap_load_module() Loading SIG from /usr/local/freeswitch/mod/ozmod_isdn.so 2009-03-14 17:44:06 [INFO] zap_io.c:2566 zap_configure_span() auto-loaded 'isdn' 2009-03-14 17:44:06 [CONSOLE] switch_loadable_module.c:863 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:142 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:240 switch_loadable_module_process() Adding Application 'disable_ec' 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:260 switch_loadable_module_process() Adding API Function 'oz' 2009-03-14 17:44:06 [ERR] Span:0 Q.921() Received UA frame in invalid state When i dial from onference: freeswitch at voipgw> conference test at moh dial openzap/1/3/81048663000000 2009-03-14 17:50:26 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel OpenZAP/1:4/81048663000000 [d6137f1a-10af-11de-8496-c9e9c714d68b] 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 2009-03-14 17:50:30 [NOTICE] mod_openzap.c:1500 on_clear_channel_signal() Ring-Ready OpenZAP/1:4/81048663000000! 2009-03-14 17:50:34 [NOTICE] mod_openzap.c:1491 on_clear_channel_signal() Pre-Answer OpenZAP/1:4/81048663000000! 2009-03-14 17:50:35 [NOTICE] mod_openzap.c:1480 on_clear_channel_signal() Channel [OpenZAP/1:4/81048663000000] has been answered API CALL [conference(test at moh dial openzap/1/4/81048663000000)] output: Call Requested: result: [SUCCESS] 2009-03-14 17:44:06 [CONSOLE] switch_loadable_module.c:863 switch_loadable_module_load_file() Successfully Loaded [mod_commands] freeswitch at voipgw> oz list API CALL [oz(list)] output: +OK span: 1 (span1) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none freeswitch at voipgw> oz dump 1 API CALL [oz(dump 1)] output: +OK span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE span_id: 1 chan_id: 2 physical_span_id: 1 physical_chan_id: 2 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE span_id: 1 chan_id: 3 physical_span_id: 1 physical_chan_id: 3 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE [...] any idea how i can fix this error ? Krzysztof Zimnicki From pereyra.roberto at gmail.com Tue Mar 17 05:26:29 2009 From: pereyra.roberto at gmail.com (Roberto Pereyra) Date: Tue, 17 Mar 2009 09:26:29 -0300 Subject: [Freeswitch-users] enable anonymous incomming calls In-Reply-To: References: Message-ID: Hi all I'm freswitch newbie ?and have a simple question. How can enable anonymous inbound calls ? I would like to use freeswitch to accept incomming calls from sipbroker DIDs Any hint ? Thank in advance for all freeswitch team !! roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar The best dedicated servers - LiquidWeb http://www.liquidweb.com/?RID=contenid From matt at hellohunter.com Tue Mar 17 05:58:24 2009 From: matt at hellohunter.com (Matt Hunter) Date: Tue, 17 Mar 2009 19:58:24 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup Message-ID: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the "fifo in" execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/7d95ee3c/attachment-0002.html From mattdfong at gmail.com Tue Mar 17 06:37:33 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 17 Mar 2009 20:37:33 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> Message-ID: <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> I apologize if this is a double post to -dev. I'm not sure why I don't see my message appearing, so I'm going to try again in the -user list (first timer posting here ;). I have a situation where it would be useful for a caller not to be hungup, after finishing the "fifo in" execution (when the agent disconnects the call or the agent hangs-up). The caller is automatically hungup, in this situation. It would be preferable if the caller channel went further along the dial plan. I thought I might get lucky implementing this setting with hangup_after_bridge to false, but fifo does not utilize this variable. I tried looking thru the mod_fifo.c source, but my c skills are minimal. I also tried executing fifo in a lua app and setting setAutoHangup(false), but that also did not work. Any chance this could be done as a feature enhancement? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/50e92e38/attachment-0002.html From dujinfang at gmail.com Tue Mar 17 07:50:02 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 17 Mar 2009 22:50:02 +0800 Subject: [Freeswitch-users] enable anonymous incomming calls In-Reply-To: References: Message-ID: <58C283C2-52C3-48E0-B124-999CB01842FC@gmail.com> at default config, in conf/sip_profiles/external.xml where $${external_sip_port} is a variable you can find in conf/ vars.xml, normally it's 5080, make sure sipbroker route calls to that port, and then you can make a dialplan in conf/dialplan/public.xml turn on verbose log in console by fs> console loglevel debug you can see the process when FS hit the dialplan On Mar 17, 2009, at 8:26 PM, Roberto Pereyra wrote: > Hi all > > I'm freswitch newbie and have a simple question. > > How can enable anonymous inbound calls ? I would like to use > freeswitch to accept incomming calls from sipbroker DIDs > > Any hint ? > > Thank in advance for all freeswitch team !! > > roberto > > -- > Ing. Roberto Pereyra > ContenidosOnline > http://www.contenidosonline.com.ar > > The best dedicated servers - LiquidWeb > http://www.liquidweb.com/?RID=contenid > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave at 3c.co.uk Tue Mar 17 08:04:29 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Mar 2009 09:04:29 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BF942A.3030305@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> Message-ID: <49BFBBFD.1050308@3c.co.uk> Steve Underwood wrote: >> When there is Echo being generated from the far end, usually in a >> bridged call. If you application is just an IVR, with no far end >> connectivity, then you shouldn't need an echo can. If you are bridging >> calls, then at some point you may need it, depending on what else is >> in the loop. >> > This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it > they give very poor reliability detecting DTMF while the prompts are > playing. If the system uses voice recognition, its reliability will be > even worse. > With respect, this is at best half true. DTMF detection has always worked just fine without echo cancellation - the Dialogic, Aculab and Rhetorex cards which I used in the late 1990s managed it perfectly well; if the DTMF detection code in * and FS can't, then maybe that's something for its author to look at ;-) ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the same hardware above back in the day. I'd be interested to see results of testing an ASR engine in with echo; unfortunately, most vendors appear to prohibit the publication of test results in their licensing. --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/061cb3cb/attachment-0002.html From pereyra.roberto at gmail.com Tue Mar 17 08:16:35 2009 From: pereyra.roberto at gmail.com (Roberto Pereyra) Date: Tue, 17 Mar 2009 12:16:35 -0300 Subject: [Freeswitch-users] enable anonymous incomming calls In-Reply-To: <58C283C2-52C3-48E0-B124-999CB01842FC@gmail.com> References: <58C283C2-52C3-48E0-B124-999CB01842FC@gmail.com> Message-ID: Thanks a lot dujinfang !! roberto 2009/3/17 dujinfang : > at default config, in conf/sip_profiles/external.xml > > ? ? > ? ? > ? ? > > where $${external_sip_port} is a variable you can find in conf/ > vars.xml, normally it's 5080, make sure sipbroker route calls to that > port, and then you can make a dialplan in > > conf/dialplan/public.xml > > turn on verbose log in console by > > fs> console loglevel debug > > you can see the process when FS hit the dialplan > > > > On Mar 17, 2009, at 8:26 PM, Roberto Pereyra wrote: > >> Hi all >> >> I'm freswitch newbie ?and have a simple question. >> >> How can enable anonymous inbound calls ? I would like to use >> freeswitch to accept incomming calls from sipbroker DIDs >> >> Any hint ? >> >> Thank in advance for all freeswitch team !! >> >> roberto >> >> -- >> Ing. Roberto Pereyra >> ContenidosOnline >> http://www.contenidosonline.com.ar >> >> The best dedicated servers - LiquidWeb >> http://www.liquidweb.com/?RID=contenid >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 best dedicated servers - LiquidWeb http://www.liquidweb.com/?RID=contenid From benke at inqnet.at Tue Mar 17 08:23:45 2009 From: benke at inqnet.at (Christian Benke) Date: Tue, 17 Mar 2009 16:23:45 +0100 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090311180742.0c6693cb@plex> References: <20090311180742.0c6693cb@plex> Message-ID: <20090317162345.75be1d2e@plex> Hi! Is this not possible with registration at a gateway or is there a other reason why i didn't get any responses on this question? Regards Christian On Wed, 11 Mar 2009 18:07:42 +0100 Christian Benke wrote: > Hi! > > I've recently started to configure a freeswitch for our new office pbx > and so far i like it very much(Coming from asterisk&openser with 2 > years experience at a ITSP. Openser was nice but i didn't like > asterisk for several reasons, so i searched for a more stable and > cleaner alternative. Freeswitch looks _very_ promising and i'd wished > i could use it for more difficult demands than a simple > office-pbx ;-)). > > So far i had little trouble(Though our installation doesn't require > much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. > > The only issue i have not resolved yet is setting the outgoing > DID("head"-number + extension, e.g. +4312345678 + 100). > > The relevant part of the default.xml looks like this atm(where > +4312345678 is our "head"-phone-number without the extensions, > ${caller_id_number} is a 3-digit extension, e.g.: 100): > > data="effective_caller_id_number=+4312345678${caller_id_number}"/> > data="sofia/gateway/sip.myisp.at/${destination_number}"/> > > I'd expect with this dialplan the effective_caller_id would be in the > "From:"-section of the INVITE, but it seems after the bridge it is > overwritten with the gateway-username i've defined in the > gateway-configuration in sip_profiles/external/. > > So instead of: > From: "Desk Phone" > ;tag=U6yQUSta2c2Xg. > i get: > From: "Desk Phone" > ;tag=U6yQUSta2c2Xg. > in the INVITE towards the sip-trunk. > > I may not have grasped yet how proper debugging with freeswitch works, > however, in the console the last action i see, before the bridge to > sofia/external is created, is the setting of the effective-caller-id, > as expected(Do you want to see the whole output?). > > I guess i don't necessarily need to register with the provider, as > they have configured the trunk for my ip-adress and i have theirs in > the ACL(inbound calls work flawless with the head-number+extension), > so maybe the registration is the reason why freeswitch does that > automatically? > > It's probably a little issue, but i don't have the overview yet to > understand how this happens, maybe someone can point me to the right > place? > > Cheers > Christian From mrene_lists at avgs.ca Tue Mar 17 08:26:18 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Mar 2009 11:26:18 -0400 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090311180742.0c6693cb@plex> References: <20090311180742.0c6693cb@plex> Message-ID: gateways have their username in the from section, callerid is sent out as remote-party-id or p-asserted-identity. if you want the from part to have the user you need to set the "caller- id-in-from" param to "true" Math On 11-Mar-09, at 1:07 PM, Christian Benke wrote: > Hi! > > I've recently started to configure a freeswitch for our new office pbx > and so far i like it very much(Coming from asterisk&openser with 2 > years experience at a ITSP. Openser was nice but i didn't like > asterisk > for several reasons, so i searched for a more stable and cleaner > alternative. Freeswitch looks _very_ promising and i'd wished i could > use it for more difficult demands than a simple office-pbx ;-)). > > So far i had little trouble(Though our installation doesn't require > much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. > > The only issue i have not resolved yet is setting the outgoing > DID("head"-number + extension, e.g. +4312345678 + 100). > > The relevant part of the default.xml looks like this atm(where > +4312345678 is our "head"-phone-number without the extensions, > ${caller_id_number} is a 3-digit extension, e.g.: 100): > > data="effective_caller_id_number=+4312345678${caller_id_number}"/> > data="sofia/gateway/sip.myisp.at/${destination_number}"/> > > I'd expect with this dialplan the effective_caller_id would be in the > "From:"-section of the INVITE, but it seems after the bridge it is > overwritten with the gateway-username i've defined in the > gateway-configuration in sip_profiles/external/. > > So instead of: > From: "Desk Phone" > ;tag=U6yQUSta2c2Xg. > i get: > From: "Desk Phone" > ;tag=U6yQUSta2c2Xg. > in the INVITE towards the sip-trunk. > > I may not have grasped yet how proper debugging with freeswitch works, > however, in the console the last action i see, before the bridge to > sofia/external is created, is the setting of the effective-caller- > id, as > expected(Do you want to see the whole output?). > > I guess i don't necessarily need to register with the provider, as > they > have configured the trunk for my ip-adress and i have theirs in > the ACL(inbound calls work flawless with the head-number+extension), > so > maybe the registration is the reason why freeswitch does that > automatically? > > It's probably a little issue, but i don't have the overview yet to > understand how this happens, maybe someone can point me to the right > place? > > Cheers > Christian > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Mar 17 08:27:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 10:27:07 -0500 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090317162345.75be1d2e@plex> References: <20090311180742.0c6693cb@plex> <20090317162345.75be1d2e@plex> Message-ID: <5B965170-FAD4-413C-B38E-FEF0834BE3FB@freeswitch.org> Try export instead of "set" /b On Mar 17, 2009, at 10:23 AM, Christian Benke wrote: >> > data="effective_caller_id_number=+4312345678${caller_id_number}"/> From steveu at coppice.org Tue Mar 17 08:36:52 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 17 Mar 2009 23:36:52 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFBBFD.1050308@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> Message-ID: <49BFC394.6070806@coppice.org> David Knell wrote: > Steve Underwood wrote: >>> When there is Echo being generated from the far end, usually in a >>> bridged call. If you application is just an IVR, with no far end >>> connectivity, then you shouldn't need an echo can. If you are bridging >>> calls, then at some point you may need it, depending on what else is >>> in the loop. >>> >> This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it >> they give very poor reliability detecting DTMF while the prompts are >> playing. If the system uses voice recognition, its reliability will be >> even worse. >> > With respect, this is at best half true. DTMF detection has always > worked just fine > without echo cancellation - the Dialogic, Aculab and Rhetorex cards > which I used > in the late 1990s managed it perfectly well; if the DTMF detection > code in * and FS > can't, then maybe that's something for its author to look at ;-) Try reading the Dialogic and Aculab documentation. Those cards used quite a bit of their DSP capability to remove the spillback of outgoing voice into their DTMF receivers. You'll find the DTMF detector in spandsp (not necessarily the ones in * or FS, which have been altered a bit) is superior to either Dialogic or Aculab's. > ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the > same > hardware above back in the day. I'd be interested to see results of > testing an ASR > engine in with echo; unfortunately, most vendors appear to prohibit > the publication > of test results in their licensing. L&H used to work fine with the J series Dialogic cards. The Dialogic documents go into considerable details about the echo cancellation arrangements to make that happen. Regards, Steve From dujinfang at gmail.com Tue Mar 17 08:44:17 2009 From: dujinfang at gmail.com (dujinfang) Date: Tue, 17 Mar 2009 23:44:17 +0800 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090317162345.75be1d2e@plex> References: <20090311180742.0c6693cb@plex> <20090317162345.75be1d2e@plex> Message-ID: <620A99C2-26DC-4575-9BF5-47725EAC89EA@gmail.com> Maybe it can help by following this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012083.html On Mar 17, 2009, at 11:23 PM, Christian Benke wrote: > Hi! > > Is this not possible with registration at a gateway or is there a > other > reason why i didn't get any responses on this question? > > Regards > Christian > > On Wed, 11 Mar 2009 18:07:42 +0100 > Christian Benke wrote: > >> Hi! >> >> I've recently started to configure a freeswitch for our new office >> pbx >> and so far i like it very much(Coming from asterisk&openser with 2 >> years experience at a ITSP. Openser was nice but i didn't like >> asterisk for several reasons, so i searched for a more stable and >> cleaner alternative. Freeswitch looks _very_ promising and i'd wished >> i could use it for more difficult demands than a simple >> office-pbx ;-)). >> >> So far i had little trouble(Though our installation doesn't require >> much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. >> >> The only issue i have not resolved yet is setting the outgoing >> DID("head"-number + extension, e.g. +4312345678 + 100). >> >> The relevant part of the default.xml looks like this atm(where >> +4312345678 is our "head"-phone-number without the extensions, >> ${caller_id_number} is a 3-digit extension, e.g.: 100): >> >> > data="effective_caller_id_number=+4312345678${caller_id_number}"/> >> > data="sofia/gateway/sip.myisp.at/${destination_number}"/> >> >> I'd expect with this dialplan the effective_caller_id would be in the >> "From:"-section of the INVITE, but it seems after the bridge it is >> overwritten with the gateway-username i've defined in the >> gateway-configuration in sip_profiles/external/. >> >> So instead of: >> From: "Desk Phone" >> ;tag=U6yQUSta2c2Xg. >> i get: >> From: "Desk Phone" >> ;tag=U6yQUSta2c2Xg. >> in the INVITE towards the sip-trunk. >> >> I may not have grasped yet how proper debugging with freeswitch >> works, >> however, in the console the last action i see, before the bridge to >> sofia/external is created, is the setting of the effective-caller-id, >> as expected(Do you want to see the whole output?). >> >> I guess i don't necessarily need to register with the provider, as >> they have configured the trunk for my ip-adress and i have theirs in >> the ACL(inbound calls work flawless with the head-number+extension), >> so maybe the registration is the reason why freeswitch does that >> automatically? >> >> It's probably a little issue, but i don't have the overview yet to >> understand how this happens, maybe someone can point me to the right >> place? >> >> Cheers >> Christian > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue Mar 17 08:48:28 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Mar 2009 08:48:28 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron wrote: > Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed." > > However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. > Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC From chris at fowler.cc Tue Mar 17 08:49:00 2009 From: chris at fowler.cc (Chris Fowler) Date: Tue, 17 Mar 2009 08:49:00 -0700 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237266076.27059.1305787305@webmail.messagingengine.com> References: <1237266076.27059.1305787305@webmail.messagingengine.com> Message-ID: <1237304940.23519.1305875953@webmail.messagingengine.com> Thanks for the tip Brian. Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log Chris. From anthony.minessale at gmail.com Tue Mar 17 08:53:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Mar 2009 10:53:05 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> Message-ID: <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> there is a patch in jira that will implement this feature about to be added 2009/3/17 Matthew Fong > I apologize if this is a double post to -dev. I'm not sure why I don't see > my message appearing, so I'm going to try again in the -user list (first > timer posting here ;). > > I have a situation where it would be useful for a caller not to be hungup, > after finishing the "fifo in" execution (when the agent disconnects the call > or the agent hangs-up). The caller is automatically hungup, in this > situation. It would be preferable if the caller channel went further along > the dial plan. I thought I might get lucky implementing this setting with > hangup_after_bridge to false, but fifo does not utilize this variable. > I tried looking thru the mod_fifo.c source, but my c skills are minimal. I > also tried executing fifo in a lua app and setting setAutoHangup(false), but > that also did not work. Any chance this could be done as a feature > enhancement? Thanks. > > --matt > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/4d60b7fb/attachment-0002.html From brian at freeswitch.org Tue Mar 17 08:53:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 10:53:02 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237304940.23519.1305875953@webmail.messagingengine.com> References: <1237266076.27059.1305787305@webmail.messagingengine.com> <1237304940.23519.1305875953@webmail.messagingengine.com> Message-ID: You're not leaking... I wouldn't call 737 bytes a leak. /b On Mar 17, 2009, at 10:49 AM, Chris Fowler wrote: > Thanks for the tip Brian. > > Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log > > Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/9ee76564/attachment-0002.html From anthony.minessale at gmail.com Tue Mar 17 08:56:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Mar 2009 10:56:50 -0500 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <20090311180742.0c6693cb@plex> References: <20090311180742.0c6693cb@plex> Message-ID: <191c3a030903170856r15ba2585x6927a00a6983d320@mail.gmail.com> The From: header is not the correct place to place the caller id in SIP yet some providers assume it is. If you add this to your gateway xml config it should fix your problem On Wed, Mar 11, 2009 at 12:07 PM, Christian Benke wrote: > Hi! > > I've recently started to configure a freeswitch for our new office pbx > and so far i like it very much(Coming from asterisk&openser with 2 > years experience at a ITSP. Openser was nice but i didn't like asterisk > for several reasons, so i searched for a more stable and cleaner > alternative. Freeswitch looks _very_ promising and i'd wished i could > use it for more difficult demands than a simple office-pbx ;-)). > > So far i had little trouble(Though our installation doesn't require > much), for PSTN-calls i'm using a SIP-Trunk provided by our ISP. > > The only issue i have not resolved yet is setting the outgoing > DID("head"-number + extension, e.g. +4312345678 + 100). > > The relevant part of the default.xml looks like this atm(where > +4312345678 is our "head"-phone-number without the extensions, > ${caller_id_number} is a 3-digit extension, e.g.: 100): > > data="effective_caller_id_number=+4312345678${caller_id_number}"/> > data="sofia/gateway/sip.myisp.at/${destination_number} > "/> > > I'd expect with this dialplan the effective_caller_id would be in the > "From:"-section of the INVITE, but it seems after the bridge it is > overwritten with the gateway-username i've defined in the > gateway-configuration in sip_profiles/external/. > > So instead of: > From: "Desk Phone" > > ;transport=udp>;tag=U6yQUSta2c2Xg. > i get: > From: "Desk Phone" > > ;transport=udp>;tag=U6yQUSta2c2Xg. > in the INVITE towards the sip-trunk. > > I may not have grasped yet how proper debugging with freeswitch works, > however, in the console the last action i see, before the bridge to > sofia/external is created, is the setting of the effective-caller-id, as > expected(Do you want to see the whole output?). > > I guess i don't necessarily need to register with the provider, as they > have configured the trunk for my ip-adress and i have theirs in > the ACL(inbound calls work flawless with the head-number+extension), so > maybe the registration is the reason why freeswitch does that > automatically? > > It's probably a little issue, but i don't have the overview yet to > understand how this happens, maybe someone can point me to the right > place? > > Cheers > Christian > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/53a093f5/attachment-0002.html From anthony.minessale at gmail.com Tue Mar 17 09:03:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Mar 2009 11:03:03 -0500 Subject: [Freeswitch-users] Possible memory / cpu leak In-Reply-To: <1237304940.23519.1305875953@webmail.messagingengine.com> References: <1237266076.27059.1305787305@webmail.messagingengine.com> <1237304940.23519.1305875953@webmail.messagingengine.com> Message-ID: <191c3a030903170903m41f67916j78f7457af32090c1@mail.gmail.com> The crash on shutdown was an issue in mod_spidermonkey that was accidentally added if you update again it's gone. please run the valgrind command again then make several calls that fall in line with your normal usage pattern so the program can get an accurate trace of the memory usage. On Tue, Mar 17, 2009 at 10:49 AM, Chris Fowler wrote: > Thanks for the tip Brian. > > Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log > > Chris. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/426e6ce9/attachment-0002.html From kzimnicki at gcdf.pl Tue Mar 17 09:24:23 2009 From: kzimnicki at gcdf.pl (Krzysztof Zimnicki) Date: Tue, 17 Mar 2009 17:24:23 +0100 Subject: [Freeswitch-users] Digium TE220P problem. Message-ID: <49BFCEB7.5010703@gcdf.pl> Hi, I have problem with Digium TE220P. Everything works, i can call & talk, but everytime i have CRIT message: 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 When FS start show me ERR message: 2009-03-14 17:44:06 [ERR] Span:0 Q.921() Received UA frame in invalid state This is my config: cat zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span = 2,0,0,ccs,hdb3,crc4 bchan = 32-46,48-62 dchan = 47 loadzone = ru defaultzone=ru cat openzap.conf [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 cat openzap.conf.xml and FS start LOGS: 2009-03-14 17:44:06 [NOTICE] zap_io.c:2612 zap_global_init() Modules configured: 1 2009-03-14 17:44:06 [NOTICE] ozmod_zt.c:922 zt_init() Using Zaptel control device 2009-03-14 17:44:06 [INFO] zap_io.c:2433 zap_load_module() Loading IO from /usr/local/freeswitch/mod/ozmod_zt.so [zt] 2009-03-14 17:44:06 [INFO] zap_io.c:2233 load_config() auto-loaded 'zt' 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:38 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 2 as OpenZAP device 1:2 fd:39 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 3 as OpenZAP device 1:3 fd:40 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 4 as OpenZAP device 1:4 fd:41 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 5 as OpenZAP device 1:5 fd:42 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 6 as OpenZAP device 1:6 fd:43 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 7 as OpenZAP device 1:7 fd:44 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 8 as OpenZAP device 1:8 fd:45 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 9 as OpenZAP device 1:9 fd:46 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 10 as OpenZAP device 1:10 fd:47 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 11 as OpenZAP device 1:11 fd:48 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 12 as OpenZAP device 1:12 fd:49 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 13 as OpenZAP device 1:13 fd:50 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 14 as OpenZAP device 1:14 fd:51 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 15 as OpenZAP device 1:15 fd:52 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 16 as OpenZAP device 1:16 fd:53 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 17 as OpenZAP device 1:17 fd:54 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 18 as OpenZAP device 1:18 fd:55 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 19 as OpenZAP device 1:19 fd:56 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 20 as OpenZAP device 1:20 fd:57 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 21 as OpenZAP device 1:21 fd:58 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 22 as OpenZAP device 1:22 fd:59 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 23 as OpenZAP device 1:23 fd:60 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 24 as OpenZAP device 1:24 fd:61 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 25 as OpenZAP device 1:25 fd:62 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 26 as OpenZAP device 1:26 fd:63 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 27 as OpenZAP device 1:27 fd:64 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 28 as OpenZAP device 1:28 fd:65 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 29 as OpenZAP device 1:29 fd:66 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 30 as OpenZAP device 1:30 fd:67 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 31 as OpenZAP device 1:31 fd:68 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 32 as OpenZAP device 2:1 fd:69 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 33 as OpenZAP device 2:2 fd:70 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 34 as OpenZAP device 2:3 fd:71 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 35 as OpenZAP device 2:4 fd:72 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 36 as OpenZAP device 2:5 fd:73 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 37 as OpenZAP device 2:6 fd:74 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 38 as OpenZAP device 2:7 fd:75 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 39 as OpenZAP device 2:8 fd:76 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 40 as OpenZAP device 2:9 fd:77 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 41 as OpenZAP device 2:10 fd:78 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 42 as OpenZAP device 2:11 fd:79 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 43 as OpenZAP device 2:12 fd:80 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 44 as OpenZAP device 2:13 fd:81 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 45 as OpenZAP device 2:14 fd:82 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 46 as OpenZAP device 2:15 fd:83 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 47 as OpenZAP device 2:16 fd:84 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 48 as OpenZAP device 2:17 fd:85 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 49 as OpenZAP device 2:18 fd:86 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 50 as OpenZAP device 2:19 fd:87 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 51 as OpenZAP device 2:20 fd:88 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 52 as OpenZAP device 2:21 fd:89 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 53 as OpenZAP device 2:22 fd:90 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 54 as OpenZAP device 2:23 fd:91 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 55 as OpenZAP device 2:24 fd:92 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 56 as OpenZAP device 2:25 fd:93 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 57 as OpenZAP device 2:26 fd:94 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 58 as OpenZAP device 2:27 fd:95 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 59 as OpenZAP device 2:28 fd:96 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 60 as OpenZAP device 2:29 fd:97 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 61 as OpenZAP device 2:30 fd:98 2009-03-14 17:44:06 [INFO] ozmod_zt.c:299 zt_open_range() configuring device /dev/zap/channel channel 62 as OpenZAP device 2:31 fd:99 2009-03-14 17:44:06 [INFO] zap_io.c:2356 load_config() Configured 62 channel(s) 2009-03-14 17:44:06 [INFO] zap_io.c:2450 zap_load_module() Loading SIG from /usr/local/freeswitch/mod/ozmod_isdn.so 2009-03-14 17:44:06 [INFO] zap_io.c:2566 zap_configure_span() auto-loaded 'isdn' 2009-03-14 17:44:06 [CONSOLE] switch_loadable_module.c:863 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:142 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:240 switch_loadable_module_process() Adding Application 'disable_ec' 2009-03-14 17:44:06 [NOTICE] switch_loadable_module.c:260 switch_loadable_module_process() Adding API Function 'oz' 2009-03-14 17:44:06 [ERR] Span:0 Q.921() Received UA frame in invalid state When i dial from onference: freeswitch at voipgw> conference test at moh dial openzap/1/3/81048663000000 2009-03-14 17:50:26 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel OpenZAP/1:4/81048663000000 [d6137f1a-10af-11de-8496-c9e9c714d68b] 2009-03-14 17:50:30 [CRIT] ozmod_isdn.c:904 zap_isdn_931_34() Received CALL PROCEEDING message for channel 0 2009-03-14 17:50:30 [NOTICE] mod_openzap.c:1500 on_clear_channel_signal() Ring-Ready OpenZAP/1:4/81048663000000! 2009-03-14 17:50:34 [NOTICE] mod_openzap.c:1491 on_clear_channel_signal() Pre-Answer OpenZAP/1:4/81048663000000! 2009-03-14 17:50:35 [NOTICE] mod_openzap.c:1480 on_clear_channel_signal() Channel [OpenZAP/1:4/81048663000000] has been answered API CALL [conference(test at moh dial openzap/1/4/81048663000000)] output: Call Requested: result: [SUCCESS] 2009-03-14 17:44:06 [CONSOLE] switch_loadable_module.c:863 switch_loadable_module_load_file() Successfully Loaded [mod_commands] freeswitch at voipgw> oz list API CALL [oz(list)] output: +OK span: 1 (span1) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none freeswitch at voipgw> oz dump 1 API CALL [oz(dump 1)] output: +OK span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE span_id: 1 chan_id: 2 physical_span_id: 1 physical_chan_id: 2 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE span_id: 1 chan_id: 3 physical_span_id: 1 physical_chan_id: 3 type: B state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE [...] any idea how i can fix this error ? Krzysztof Zimnicki From tntknight at gmail.com Tue Mar 17 09:24:58 2009 From: tntknight at gmail.com (Anthony Knight) Date: Tue, 17 Mar 2009 12:24:58 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFC394.6070806@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> Message-ID: <4cd9d780903170924m79e637d4y4f715891ce16a663@mail.gmail.com> Thanks for the feedback. I have plenty of experience with IVRs and Dialogic cards (starting with D121/LSI120s and SS96s under DOS in the 90's all the way up to Intel's DM/Vs) and didn't ever have a problem with DTMF collection with ISDN PRI lines except occasionally with wireless and cell phones (Bad line quality). These new cards are so much cheaper than the Dialogic cards were, I should just buy the version with the cancellers. Tony On Tue, Mar 17, 2009 at 11:36 AM, Steve Underwood wrote: > David Knell wrote: > > Steve Underwood wrote: > >>> When there is Echo being generated from the far end, usually in a > >>> bridged call. If you application is just an IVR, with no far end > >>> connectivity, then you shouldn't need an echo can. If you are bridging > >>> calls, then at some point you may need it, depending on what else is > >>> in the loop. > >>> > >> This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it > >> they give very poor reliability detecting DTMF while the prompts are > >> playing. If the system uses voice recognition, its reliability will be > >> even worse. > >> > > With respect, this is at best half true. DTMF detection has always > > worked just fine > > without echo cancellation - the Dialogic, Aculab and Rhetorex cards > > which I used > > in the late 1990s managed it perfectly well; if the DTMF detection > > code in * and FS > > can't, then maybe that's something for its author to look at ;-) > Try reading the Dialogic and Aculab documentation. Those cards used > quite a bit of their DSP capability to remove the spillback of outgoing > voice into their DTMF receivers. You'll find the DTMF detector in > spandsp (not necessarily the ones in * or FS, which have been altered a > bit) is superior to either Dialogic or Aculab's. > > ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the > > same > > hardware above back in the day. I'd be interested to see results of > > testing an ASR > > engine in with echo; unfortunately, most vendors appear to prohibit > > the publication > > of test results in their licensing. > L&H used to work fine with the J series Dialogic cards. The Dialogic > documents go into considerable details about the echo cancellation > arrangements to make that happen. > > Regards, > Steve > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/2458886b/attachment-0002.html From benke at inqnet.at Tue Mar 17 09:38:35 2009 From: benke at inqnet.at (Christian Benke) Date: Tue, 17 Mar 2009 17:38:35 +0100 Subject: [Freeswitch-users] bridge to gateway overwrites "effective caller id" with "username" In-Reply-To: <191c3a030903170856r15ba2585x6927a00a6983d320@mail.gmail.com> References: <20090311180742.0c6693cb@plex> <191c3a030903170856r15ba2585x6927a00a6983d320@mail.gmail.com> Message-ID: <20090317173835.68ee5df3@plex> wow, now that was fast :-) Cheers for all replies, setting the caller-id-in-from-parameter was sufficient! regards Christian From steveu at coppice.org Tue Mar 17 09:38:55 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 00:38:55 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <4cd9d780903170924m79e637d4y4f715891ce16a663@mail.gmail.com> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <4cd9d780903170924m79e637d4y4f715891ce16a663@mail.gmail.com> Message-ID: <49BFD21F.3060303@coppice.org> Anthony Knight wrote: > Thanks for the feedback. > > I have plenty of experience with IVRs and Dialogic cards (starting > with D121/LSI120s and SS96s under DOS in the 90's all the way up to > Intel's DM/Vs) and didn't ever have a problem with DTMF collection > with ISDN PRI lines except occasionally with wireless and cell phones > (Bad line quality). You have my deepest sympathy. :-) Steve From msc at freeswitch.org Tue Mar 17 09:42:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Mar 2009 09:42:05 -0700 Subject: [Freeswitch-users] [OpenZap] problem with TE220P In-Reply-To: <49BF6A99.6070308@gcdf.pl> References: <49BF6A99.6070308@gcdf.pl> Message-ID: <87f2f3b90903170942m3ae6bf05x1a827ec49efcf46a@mail.gmail.com> > any idea how i can fix this error ? > I believe this is a harmless warning. However, you might try to use ozmod_libpri, which uses the libpri PRI stack instead of the built-in OpenZAP PRI stack. More info here: http://wiki.freeswitch.org/wiki/OpenZAP#OpenZAP_Installation -MC From dave at 3c.co.uk Tue Mar 17 10:46:27 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Mar 2009 11:46:27 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFC394.6070806@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> Message-ID: <49BFE1F3.2030207@3c.co.uk> Steve Underwood wrote: > David Knell wrote: > >> Steve Underwood wrote: >> >>>> When there is Echo being generated from the far end, usually in a >>>> bridged call. If you application is just an IVR, with no far end >>>> connectivity, then you shouldn't need an echo can. If you are bridging >>>> calls, then at some point you may need it, depending on what else is >>>> in the loop. >>>> >>>> >>> This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it >>> they give very poor reliability detecting DTMF while the prompts are >>> playing. If the system uses voice recognition, its reliability will be >>> even worse. >>> >>> >> With respect, this is at best half true. DTMF detection has always >> worked just fine >> without echo cancellation - the Dialogic, Aculab and Rhetorex cards >> which I used >> in the late 1990s managed it perfectly well; if the DTMF detection >> code in * and FS >> can't, then maybe that's something for its author to look at ;-) >> > Try reading the Dialogic and Aculab documentation. Those cards used > quite a bit of their DSP capability to remove the spillback of outgoing > voice into their DTMF receivers. You'll find the DTMF detector in > spandsp (not necessarily the ones in * or FS, which have been altered a > bit) is superior to either Dialogic or Aculab's. > The first bit of that's a tad patronising, isn't it, and, in the case of the decade-old Aculab cards which which I'm most familiar, is also untrue. As for the second, do you have any test results to back that up? I'm more curious than setting out for an argument.. >> ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the >> same >> hardware above back in the day. I'd be interested to see results of >> testing an ASR >> engine in with echo; unfortunately, most vendors appear to prohibit >> the publication >> of test results in their licensing. >> > L&H used to work fine with the J series Dialogic cards. The Dialogic > documents go into considerable details about the echo cancellation > arrangements to make that happen. > > You've missed the point I was trying to make. It used to work fine with no echo cancellation at all. --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/5ccc6fae/attachment-0002.html From ludovic.fouquet at bewan.com Tue Mar 17 11:16:33 2009 From: ludovic.fouquet at bewan.com (ludovic) Date: Tue, 17 Mar 2009 19:16:33 +0100 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: References: <49BE928D.3090509@bewan.com> <49BF87E5.5090809@bewan.com> Message-ID: <49BFE901.3070709@bewan.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/faa72201/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: bewan100.jpg Type: image/jpeg Size: 3963 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/faa72201/attachment-0002.jpg From brian at freeswitch.org Tue Mar 17 11:26:31 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 13:26:31 -0500 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: <49BFE901.3070709@bewan.com> References: <49BE928D.3090509@bewan.com> <49BF87E5.5090809@bewan.com> <49BFE901.3070709@bewan.com> Message-ID: <246296A5-851A-4859-BCA9-05E2415A20EA@freeswitch.org> I checked, I don't see sw1.freephonie.net in the logs trying to resolve it... and the SRV records are all correct as are the naptr records which is shocking ;) /b On Mar 17, 2009, at 1:16 PM, ludovic wrote: > I understood the example. > What I mean is that my DNS issue comes from sofia-sip and my sip > provider (freephonie.net) name resolution which fails when calling > whereas it is well resolved during the registration process. > Here is a trace : > 2009-03-17 18:47:28 [NOTICE] switch_channel.c:538 > switch_channel_set_name() New Channel sofia/external/0123456789 > [ae9c2108-131b-11de-8a2f-1fafbaa54120] nua: > nh_create_handle: entering > ; > nua: nua_handle_bind: entering > nua: nua_invite: entering > nua: nua_stack_set_params: entering > soa_clone(static::0x100abd98, 0x100a8738, 0x100c6748) called > soa_set_params(static::0x100b4728, ...) called > soa_set_params(static::0x100b4728, ...) called > soa_set_user_sdp(static::0x100b4728, (nil), 0x10106364, -1) called > soa_set_capability_sdp(static::0x100b4728, (nil), 0x10106364, -1) > called > su_localinfo: if lo with index 1 > su_localinfo: if lan1 with index 18 > su_localinfo: if ppp1 with index 23 > nta_leg_tcreate(0x100c7f30) > nua(0x100c6748): adding session usage > soa_init_offer_answer(static::0x100b4728) called > soa_generate_offer(static::0x100b4728, 0) called > soa_static_offer_answer_action(0x100b4728, soa_generate_offer): called > soa_static(0x100b4728, soa_generate_offer): generating local > description > su_localinfo: if lo with index 1 > su_localinfo: if lan1 with index 18 > su_localinfo: if ppp1 with index 23 > soa_static(0x100b4728, soa_generate_offer): upgrade with local > description > soa_sdp_mode_set(0x7dbfd850, (nil), ""): called > soa_static(0x100b4728, soa_generate_offer): storing local description > soa_get_local_sdp(static::0x100b4728, [(nil)], [0x7dbff980], > [0x7dbff984]) called > nta: selecting scheme sip > sres_cache_get(0x100abfc0, NAPTR, "freephonie.net.") called > rr found in cache: freephonie.net. 35 > sres_cache_get(0x100abfc0, NAPTR, "freephonie.net.") returned 1 > entries > nta: for "freephonie.net" query "freephonie.net" NAPTR (cached) > nta: freephonie.net. IN NAPTR 100 100 "s" "SIP+D2U" "" > _sip_udp.freephonie.net. > sres_cache_get(0x100abfc0, SRV, "_sip_udp.freephonie.net.") called > rr found in cache: _sip_udp.freephonie.net. 33 > sres_cache_get(0x100abfc0, SRV, "_sip_udp.freephonie.net.") returned > 1 entries > nta: for "freephonie.net" query "_sip_udp.freephonie.net." SRV > (cached) > nta: timer set to 32000 ms > nua(0x100c6748): call state changed: init -> calling, sent offer > soa_get_local_sdp(static::0x100b4728, [0x7dbff988], [0x7dbff98c], > [(nil)]) called > nua: nua_application_event: entering > nua(0x100c6748): call state changed: calling -> init > nua(0x100c6748): removing session usage > soa_destroy(static::0x100b4728) called From steveu at coppice.org Tue Mar 17 11:28:27 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 02:28:27 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFE1F3.2030207@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> Message-ID: <49BFEBCB.9020708@coppice.org> David Knell wrote: > Steve Underwood wrote: >> David Knell wrote: >> >>> Steve Underwood wrote: >>> >>>>> When there is Echo being generated from the far end, usually in a >>>>> bridged call. If you application is just an IVR, with no far end >>>>> connectivity, then you shouldn't need an echo can. If you are bridging >>>>> calls, then at some point you may need it, depending on what else is >>>>> in the loop. >>>>> >>>>> >>>> This is VERY VERY WRONG. IVRs badly need echo cancellation. Without it >>>> they give very poor reliability detecting DTMF while the prompts are >>>> playing. If the system uses voice recognition, its reliability will be >>>> even worse. >>>> >>>> >>> With respect, this is at best half true. DTMF detection has always >>> worked just fine >>> without echo cancellation - the Dialogic, Aculab and Rhetorex cards >>> which I used >>> in the late 1990s managed it perfectly well; if the DTMF detection >>> code in * and FS >>> can't, then maybe that's something for its author to look at ;-) >>> >> Try reading the Dialogic and Aculab documentation. Those cards used >> quite a bit of their DSP capability to remove the spillback of outgoing >> voice into their DTMF receivers. You'll find the DTMF detector in >> spandsp (not necessarily the ones in * or FS, which have been altered a >> bit) is superior to either Dialogic or Aculab's. >> > The first bit of that's a tad patronising, isn't it, You are the one who started out being offensive. > and, in the case of the decade-old Aculab > cards which which I'm most familiar, is also untrue. I can't find too much about the old cards on the web now, but I found http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html which is pretty much a copy and paste from the old Dialogic web pages, and you'll see it says "Cut through : Local echo cancellation permits 100% detection with a >4.5 dB return loss line". The Aculabs did the same thing for sure. They just couldn't work without cancellation. There were some very early Dialogic cards, using DTMF receiver chips and OKI ADPCM chips, and had no general purpose DSPs. They performed really badly because of the lack of cancellation, and were quickly replaced with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms into a Motorola 56k DSP chips. > > As for the second, do you have any test results to back that up? I'm > more curious than > setting out for an argument.. >>> ASR - yes, maybe, but L&H's ASR1500 used to work perfectly well on the >>> same >>> hardware above back in the day. I'd be interested to see results of >>> testing an ASR >>> engine in with echo; unfortunately, most vendors appear to prohibit >>> the publication >>> of test results in their licensing. >>> >> L&H used to work fine with the J series Dialogic cards. The Dialogic >> documents go into considerable details about the echo cancellation >> arrangements to make that happen. >> >> > You've missed the point I was trying to make. It used to work fine > with no echo cancellation > at all. I think you've missed the point. These things don't work by pixey dust. They work by engineering. If you have any old J or JCT cards around record the signal from the far end. You'll find only the tiniest trace of the outgoing signal mixed in with it. How do you think that happens? Steve From jp.manchu at gmail.com Tue Mar 17 13:48:18 2009 From: jp.manchu at gmail.com (JayaPrakash) Date: Wed, 18 Mar 2009 02:18:18 +0530 Subject: [Freeswitch-users] Nibblebill - DB Error while updating cash! Message-ID: Hi All, I have installed nibblebill and* it is able to bill the calls.* However, it is giving following error in FreeSwitch server. 2009-03-17 23:17:19 [DEBUG] mod_nibblebill.c:283 bill_event() Doing update query [UPDATE accounts SET cash=cash-0.045767 WHERE id="1"] 2009-03-17 23:17:19 [CRIT] mod_nibblebill.c:286 bill_event() DB Error while updating cash! Thanks Jayaprakash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/f95a7def/attachment-0002.html From ctalle at voiceway.ca Tue Mar 17 14:13:16 2009 From: ctalle at voiceway.ca (Cristian Talle) Date: Tue, 17 Mar 2009 17:13:16 -0400 Subject: [Freeswitch-users] DTMF detection during bridge Message-ID: <49C0126C.2080107@voiceway.ca> Hi, Is there any easy way to get in FS the same behavior as when using the "d" flag with asterisk's Dial command? I need FS to jump to a different extension if the caller presses a digit while waiting for the called party to answer. *"...d*: intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered..." Thanks, Cristian From brian at freeswitch.org Tue Mar 17 14:24:53 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 16:24:53 -0500 Subject: [Freeswitch-users] DTMF detection during bridge In-Reply-To: <49C0126C.2080107@voiceway.ca> References: <49C0126C.2080107@voiceway.ca> Message-ID: Check out the bind_meta_app that exists in the default examples... I think thats what you want. /b On Mar 17, 2009, at 4:13 PM, Cristian Talle wrote: > Hi, > > Is there any easy way to get in FS the same behavior as when using the > "d" flag with asterisk's Dial command? > I need FS to jump to a different extension if the caller presses a > digit > while waiting for the called party to answer. > > *"...d*: intercepts any dtmf while waiting for the call to be answered > and returns that value on the spot. This allows you to dial a 1-digit > exit extension while waiting for the call to be answered..." > > Thanks, > Cristian From mthomas at themarketbuilder.com Tue Mar 17 15:35:48 2009 From: mthomas at themarketbuilder.com (Mark Thomas) Date: Tue, 17 Mar 2009 15:35:48 -0700 Subject: [Freeswitch-users] Newbie question: Why can't I dial? Message-ID: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> Hello, everyone. I am new to Freeswitch, and telephony in general. I am trying to set up a Freeswitch system at work for a project, and I have hit a wall. I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I believe I have openzap correctly installed in wanpipe mode. I am trying to bridge an incoming SIP call from an IP phone to an openzap channel without success. The Freeswitch log shows that dialing takes place, but the call never completes. The call log is here: http://pastebin.freeswitch.org/7805 The dialplan xml, openzap.conf, and openzap.conf.xml are here: http://pastebin.freeswitch.org/7806 Any help greatly appreciated. --Mark From brian at freeswitch.org Tue Mar 17 15:54:47 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 17:54:47 -0500 Subject: [Freeswitch-users] Newbie question: Why can't I dial? In-Reply-To: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> References: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> Message-ID: Mark, You should join both #openzap and #freeswitch on irc.freenode.net there are way too many things to go over and the list would just be too slow. /b On Mar 17, 2009, at 5:35 PM, Mark Thomas wrote: > Hello, everyone. > > I am new to Freeswitch, and telephony in general. I am trying to > set up a Freeswitch system at work for a project, and I have hit a > wall. > > I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I > believe I have openzap correctly installed in wanpipe mode. I am > trying to bridge an incoming SIP call from an IP phone to an openzap > channel without success. The Freeswitch log shows that dialing > takes place, but the call never completes. > > The call log is here: http://pastebin.freeswitch.org/7805 > The dialplan xml, openzap.conf, and openzap.conf.xml are here: http://pastebin.freeswitch.org/7806 > > Any help greatly appreciated. > > --Mark > > ______________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/793c969f/attachment-0002.html From msc at freeswitch.org Tue Mar 17 16:17:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Mar 2009 16:17:05 -0700 Subject: [Freeswitch-users] Newbie question: Why can't I dial? In-Reply-To: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> References: <6B96ACC4F309A44B99EF6C795287F5AF4361891067@exchange.mbidomain.com> Message-ID: <87f2f3b90903171617l266db132vf9143c4562cad407@mail.gmail.com> On Tue, Mar 17, 2009 at 3:35 PM, Mark Thomas wrote: > Hello, everyone. > > I am new to Freeswitch, and telephony in general. ?I am trying to set up a Freeswitch system at work for a project, and I have hit a wall. > > I have a dedicated LD T1 from Qwest and a Sangoma A104 card. ?I believe I have openzap correctly installed in wanpipe mode. I am trying to bridge an incoming SIP call from an IP phone to an openzap channel without success. ?The Freeswitch log shows that dialing takes place, but the call never completes. > > The call log is here: http://pastebin.freeswitch.org/7805 > The dialplan xml, openzap.conf, and openzap.conf.xml are here: http://pastebin.freeswitch.org/7806 > > Any help greatly appreciated. > Actually I found two things you need to change in the dialplan. What's happening is that you are telling openzap to dial out span 1, lowest channel number, but you don't actually give it a phone number to dial. Here's the current dialplan: first, your expression is a bit dangerous. second, it doesn't actually "capture" the dialed number. I recommend that you do something like this: Note the leading nine, the \d+ and the parentheses. Essentially this regex says: Match any string of digits that begins with a 9 and has at least one additional digit. The parens will put the value of (\d+) into the variable $1. Your bridge then would be this: Now, reload your dialplan (press F6 or type "reloadxml" at the CLI) and dial out with a leading 9: 95551212 will send 5551212 to the telco. Try it and report back! -MC (IRC: mercutioviz) > --Mark > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dave at 3c.co.uk Tue Mar 17 16:21:11 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Mar 2009 17:21:11 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49BFEBCB.9020708@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> Message-ID: <49C03067.7070406@3c.co.uk> Steve Underwood wrote: > [whopping big snip] > >> The first bit of that's a tad patronising, isn't it, >> > You are the one who started out being offensive. > I'm sorry if you find disagreement offensive; you might not wish to read beyond this point if so. >> and, in the case of the decade-old Aculab >> cards which which I'm most familiar, is also untrue. >> > I can't find too much about the old cards on the web now, but I found > http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html > which is pretty much a copy and paste from the old Dialogic web pages, > and you'll see it says "Cut through : Local echo cancellation permits > 100% detection with a >4.5 dB return loss line". The Aculabs did the > same thing for sure. They just couldn't work without cancellation. There > were some very early Dialogic cards, using DTMF receiver chips and OKI > ADPCM chips, and had no general purpose DSPs. They performed really > badly because of the lack of cancellation, and were quickly replaced > with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms > into a Motorola 56k DSP chips. > The same document, under the bit which you've quoted, says: "(E-1) Digital trunks use separate transmit and receive paths to network. Performance dependent on far end handset's match to local analog loop." - i.e. the card does no echo cancellation. Aculab didn't even offer echo cancellation on Prosody for years and, when they did, it consumed prodigious amounts of DSP. Nonetheless, the DTMF detection worked perfectly well, even across 120 channels per 40MHz SHARC - there's just no way that those DSPs had enough horsepower to do echo cancellation across that many channels. An Asterisk box with an el-cheapo quad E1 card in that I use for TDM-SIP gatewaying detects DTMF perfectly well with no echo cancellation. You just don't need echo cancellation to achieve perfectly acceptable DTMF detection. ASR - yes, maybe, but surely only in the case where the application requires barge-in; even then, I'd be interested to see some test results, particuarly where the outbound prompt is killed the moment the ASR reports start of speech. I'm afraid that your original bald claim - that "IVRs badly need echo cancellation" is simply wrong, misleading and irresponsible: those believing it will end up spending large sums of money on technology which they probably do not need. --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/a3e1397f/attachment-0002.html From msc at freeswitch.org Tue Mar 17 16:40:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Mar 2009 16:40:38 -0700 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C03067.7070406@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> Message-ID: <87f2f3b90903171640i25da1894xa062d4f91450e97f@mail.gmail.com> > I'm afraid that your original bald claim - that "IVRs badly need echo > cancellation" is simply > wrong, misleading and irresponsible: those believing it will end up spending > large sums > of money on technology which they probably do not need. Anybody with years, perhaps decades, of DSP programming experience plus testing in the real world - and all over the world - has my vote of confidence. Furthermore, when this person writes spandsp, makes it open source, and freely answers questions about it on public fora, I am inclined not only to believe him but to trust his judgment. Bottom line: thousands of people have chosen to heed Steve's advice. He is well-respected in many technical communities. His reputation is as solid as it gets. "Do what Steve says" is about the safest bet you will ever make in this business. -MC From steveu at coppice.org Tue Mar 17 17:25:01 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 08:25:01 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C03067.7070406@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> Message-ID: <49C03F5D.9050904@coppice.org> David Knell wrote: > Steve Underwood wrote: >> [whopping big snip] >> >>> The first bit of that's a tad patronising, isn't it, >>> >> You are the one who started out being offensive. >> > I'm sorry if you find disagreement offensive; you might not wish to > read beyond this > point if so. >>> and, in the case of the decade-old Aculab >>> cards which which I'm most familiar, is also untrue. >>> >> I can't find too much about the old cards on the web now, but I found >> http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html >> which is pretty much a copy and paste from the old Dialogic web pages, >> and you'll see it says "Cut through : Local echo cancellation permits >> 100% detection with a >4.5 dB return loss line". The Aculabs did the >> same thing for sure. They just couldn't work without cancellation. There >> were some very early Dialogic cards, using DTMF receiver chips and OKI >> ADPCM chips, and had no general purpose DSPs. They performed really >> badly because of the lack of cancellation, and were quickly replaced >> with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms >> into a Motorola 56k DSP chips. >> > The same document, under the bit which you've quoted, says: > "(E-1) Digital trunks use separate transmit and receive paths to network. > Performance dependent on far end handset's match to local analog loop." > - i.e. the card does no echo cancellation. Your messages are starting to looked deranged. Why would they only apply echo cancellation to T1s? Its a bizarre idea, and you must realise its wrong. Are you so desperate to support a wrong answer you'll clutch at straws? :-\ > > Aculab didn't even offer echo cancellation on Prosody for years and, > when they did, it > consumed prodigious amounts of DSP. Nonetheless, the DTMF detection > worked > perfectly well, even across 120 channels per 40MHz SHARC - there's > just no way > that those DSPs had enough horsepower to do echo cancellation across > that many > channels. This page http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html seems to support what you say. It also implies DTMF detection sucks unless you echo cancel. The statement "If the outgoing signal is a tone of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" is telling you to keep your outgoing signal in the same frequency range as dial-tone where the dial-tone filter on the DTMF receiver will obviate the need for an echo canceller. They are freely admitting exactly what I have said. If you want a normal IVR with cut-through to work you better turn that echo canceller on. My only experience with Aculab was fitting a box designed by other people into a system. That one definitely echo cancelled, as it worked as well as the Dialogic based boxes we developed ourselves. > > An Asterisk box with an el-cheapo quad E1 card in that I use for > TDM-SIP gatewaying > detects DTMF perfectly well with no echo cancellation. You must have very low standards for "works well". > > You just don't need echo cancellation to achieve perfectly acceptable > DTMF detection. Well, not if you expect people to wait for silence before entering DTMF, but who would tolerate that these days? Cut through has been de rigeur since the late 80s. > > ASR - yes, maybe, but surely only in the case where the application > requires barge-in; > even then, I'd be interested to see some test results, particuarly > where the outbound prompt > is killed the moment the ASR reports start of speech. Doesn't any sane system expect barge in to be nearly as reliable as waiting for silence? Who would tolerate something that doesn't? It has been a standard expectation of any decent IVR since they began. > > I'm afraid that your original bald claim - that "IVRs badly need echo > cancellation" is simply > wrong, misleading and irresponsible: those believing it will end up > spending large sums > of money on technology which they probably do not need. You must have very low standards for what works well. If you suggest people leave out echo cancellation you are just asking for customer service issues down the line. That whole Aculab page is a clear response to just such issues they had, which forced them to add the necessary improvements. Regards, Steve From d at d-man.org Tue Mar 17 18:31:04 2009 From: d at d-man.org (Darren Schreiber) Date: Tue, 17 Mar 2009 18:31:04 -0700 Subject: [Freeswitch-users] Nibblebill - DB Error while updating cash! In-Reply-To: References: Message-ID: Hi there, The updates to the DB are working, but the error is still being thrown. I will try and fix this tonight. Diego also reported the same issue last week, I just haven't gotten around to it. My apologies. The bug is filed, I'll close it out when it's fixed. - Darren _____ From: JayaPrakash [mailto:jp.manchu at gmail.com] Sent: Tuesday, March 17, 2009 1:48 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Nibblebill - DB Error while updating cash! Hi All, I have installed nibblebill and it is able to bill the calls. However, it is giving following error in FreeSwitch server. 2009-03-17 23:17:19 [DEBUG] mod_nibblebill.c:283 bill_event() Doing update query [UPDATE accounts SET cash=cash-0.045767 WHERE id="1"] 2009-03-17 23:17:19 [CRIT] mod_nibblebill.c:286 bill_event() DB Error while updating cash! Thanks Jayaprakash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/7190920b/attachment-0002.html From jason at jasonjgw.net Tue Mar 17 18:46:59 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 18 Mar 2009 12:46:59 +1100 Subject: [Freeswitch-users] TLS support in Debian build Message-ID: <20090318014659.GA15840@jdc.jasonjgw.net> I've just tried enabling TLS support, and the SIP profiles with TLS enabled in them won't load. According to the wiki, this is typically the result of missing headers during the build process, with TLS having not been included. However, on my Debian system, I have header files under /usr/include/openssl, which come from the libssl-dev package. Thus, SSL/TLS should have been compiled into FreeSWITCH unless there's something amiss with the Debian build process. Suggestions for tracking this down are welcome. There's no urgency: this is just for experimental purposes, after all. From brian at freeswitch.org Tue Mar 17 18:53:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Mar 2009 20:53:58 -0500 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: <20090318014659.GA15840@jdc.jasonjgw.net> References: <20090318014659.GA15840@jdc.jasonjgw.net> Message-ID: <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> if you installed the ssl devel stuff AFTER you configured you'll need to reconfigure. /b On Mar 17, 2009, at 8:46 PM, Jason White wrote: > I've just tried enabling TLS support, and the SIP profiles with TLS > enabled in > them won't load. > > According to the wiki, this is typically the result of missing > headers during > the build process, with TLS having not been included. > > However, on my Debian system, I have header files under /usr/include/ > openssl, > which come from the libssl-dev package. Thus, SSL/TLS should have been > compiled into FreeSWITCH unless there's something amiss with the > Debian build > process. > > Suggestions for tracking this down are welcome. There's no urgency: > this is > just for experimental purposes, after all. From dave at 3c.co.uk Tue Mar 17 19:02:41 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Mar 2009 20:02:41 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C03F5D.9050904@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> Message-ID: <49C05641.7070309@3c.co.uk> Steve Underwood wrote: > David Knell wrote: > >> Steve Underwood wrote: >> >>> [whopping big snip] >>> >>> >>>> The first bit of that's a tad patronising, isn't it, >>>> >>>> >>> You are the one who started out being offensive. >>> >>> >> I'm sorry if you find disagreement offensive; you might not wish to >> read beyond this >> point if so. >> >>>> and, in the case of the decade-old Aculab >>>> cards which which I'm most familiar, is also untrue. >>>> >>>> >>> I can't find too much about the old cards on the web now, but I found >>> http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html >>> which is pretty much a copy and paste from the old Dialogic web pages, >>> and you'll see it says "Cut through : Local echo cancellation permits >>> 100% detection with a >4.5 dB return loss line". The Aculabs did the >>> same thing for sure. They just couldn't work without cancellation. There >>> were some very early Dialogic cards, using DTMF receiver chips and OKI >>> ADPCM chips, and had no general purpose DSPs. They performed really >>> badly because of the lack of cancellation, and were quickly replaced >>> with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms >>> into a Motorola 56k DSP chips. >>> >>> >> The same document, under the bit which you've quoted, says: >> "(E-1) Digital trunks use separate transmit and receive paths to network. >> Performance dependent on far end handset's match to local analog loop." >> - i.e. the card does no echo cancellation. >> > Your messages are starting to looked deranged. Why would they only apply > echo cancellation to T1s? Its a bizarre idea, and you must realise its > wrong. Are you so desperate to support a wrong answer you'll clutch at > straws? :-\ > More insults. Answer me this: if there were echo cancellation in use, why would DTMF detection performance depend on the far-end handset's match to the loop? And the follow-up question (which you've already pretty much asked) - if the card doesn't echo cancel for E1s, why would it for T1s? As an aside, I'm not convinced that the document's not talking about return loss on the T1 line itself, the implication being that the T1 is being carried on a single pair, which makes the first sentence about E1s make a bit more sense. But that's just a guess. >> Aculab didn't even offer echo cancellation on Prosody for years and, >> when they did, it >> consumed prodigious amounts of DSP. Nonetheless, the DTMF detection >> worked >> perfectly well, even across 120 channels per 40MHz SHARC - there's >> just no way >> that those DSPs had enough horsepower to do echo cancellation across >> that manychannels. >> > This page > http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html > seems to support what you say. It also implies DTMF detection sucks > unless you echo cancel. The statement "If the outgoing signal is a tone > of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" > is telling you to keep your outgoing signal in the same frequency range > as dial-tone where the dial-tone filter on the DTMF receiver will > obviate the need for an echo canceller. They are freely admitting > exactly what I have said. If you want a normal IVR with cut-through to > work you better turn that echo canceller on. > > My only experience with Aculab was fitting a box designed by other > people into a system. That one definitely echo cancelled, as it worked > as well as the Dialogic based boxes we developed ourselves. > That only holds true if your premise - that you need echo cancellation for good DTMF detection - is correct, which I don't believe it is. >> An Asterisk box with an el-cheapo quad E1 card in that I use for >> TDM-SIP gatewaying >> detects DTMF perfectly well with no echo cancellation. >> > You must have very low standards for "works well". > Nothing like a good old ad hominem attack. Beats reasoned argument any day. >> You just don't need echo cancellation to achieve perfectly acceptable >> DTMF detection. >> > Well, not if you expect people to wait for silence before entering DTMF, > but who would tolerate that these days? Cut through has been de rigeur > since the late 80s. > Oh, for pity's sake, you get perfectly good cut through without echo cancellation. Humour me and draw a quick mental picture of the spectrum of a random bit of speech at -20dBm; now add tones at -10dBm and -7dBm. They stick out like a pair of sore thumbs. I'm sure it's quite possible to come up with a pathological case - e.g. cut-through against a 1kHz milliwatt tone, but that sort of thing just doesn't happen in real- life IVR applications. >> ASR - yes, maybe, but surely only in the case where the application >> requires barge-in; >> even then, I'd be interested to see some test results, particuarly >> where the outbound prompt >> is killed the moment the ASR reports start of speech. >> > Doesn't any sane system expect barge in to be nearly as reliable as > waiting for silence? Who would tolerate something that doesn't? It has > been a standard expectation of any decent IVR since they began. > Sorry - ASR with barge-in has been a standard expectation since the first IVRs? >> I'm afraid that your original bald claim - that "IVRs badly need echo >> cancellation" is simply >> wrong, misleading and irresponsible: those believing it will end up >> spending large sums >> of money on technology which they probably do not need. >> > You must have very low standards for what works well. If you suggest > people leave out echo cancellation you are just asking for customer > service issues down the line. That whole Aculab page is a clear response > to just such issues they had, which forced them to add the necessary > improvements. > Repeating you ad-hominem really doesn't make it any stronger, I'm afraid. And the Aculab page you refer to offers four solutions for problems caused by far- end echo, of which cancellation is just one; not playing a stationary tone above 600Hz is another. Do you have any real-world samples of DTMF+echo which give your DTMF detection code trouble? --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/0ed589c6/attachment-0002.html From jason at jasonjgw.net Tue Mar 17 19:31:54 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 18 Mar 2009 13:31:54 +1100 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> References: <20090318014659.GA15840@jdc.jasonjgw.net> <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> Message-ID: <20090318023154.GA16523@jdc.jasonjgw.net> Brian West wrote: > if you installed the ssl devel stuff AFTER you configured you'll need > to reconfigure. I'm reasonably sure it was installed already, unless it was pulled in recently by a package upgrade. The configure script needs to look in /usr/include/openssl for the headers. I'll have a look at config.log and try to work out what it looked for and why it didn't find it. From egghunt at gmail.com Tue Mar 17 19:40:59 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Tue, 17 Mar 2009 23:40:59 -0300 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C05641.7070309@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> Message-ID: Sharing my humble experience: in Brazil we usually need echo cancellation to have reliable DTMF detection _and_ voice quality over E1 lines (be it on MFC/R2 - r2d - or ISDN PRI lines), either for sip/tdm gateway devices or IVR applications. Usually there's no need for echo cancellation on links from some Telcos, in some specific places. But we need it in the majority of cases, even when my box is just a gateway between legacy pbxes. This represents just a subset of the available E1s in the world and it's just a practical experience, but it's a fact for me. If I don't have a card with echo cancellation, I don't offer reliability to my customer; I've done that in the past and didn't work out. I'm not theoretically discussing anything, just sharing what I've been through in the last 4 or 5 years. 2009/3/17 David Knell > Steve Underwood wrote: > > David Knell wrote: > > > Steve Underwood wrote: > > > [whopping big snip] > > > > The first bit of that's a tad patronising, isn't it, > > > > You are the one who started out being offensive. > > > > I'm sorry if you find disagreement offensive; you might not wish to > read beyond this > point if so. > > > and, in the case of the decade-old Aculab > cards which which I'm most familiar, is also untrue. > > > > I can't find too much about the old cards on the web now, but I found http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html > which is pretty much a copy and paste from the old Dialogic web pages, > and you'll see it says "Cut through : Local echo cancellation permits > 100% detection with a >4.5 dB return loss line". The Aculabs did the > same thing for sure. They just couldn't work without cancellation. There > were some very early Dialogic cards, using DTMF receiver chips and OKI > ADPCM chips, and had no general purpose DSPs. They performed really > badly because of the lack of cancellation, and were quickly replaced > with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms > into a Motorola 56k DSP chips. > > > > The same document, under the bit which you've quoted, says: > "(E-1) Digital trunks use separate transmit and receive paths to network. > Performance dependent on far end handset's match to local analog loop." > - i.e. the card does no echo cancellation. > > > Your messages are starting to looked deranged. Why would they only apply > echo cancellation to T1s? Its a bizarre idea, and you must realise its > wrong. Are you so desperate to support a wrong answer you'll clutch at > straws? :-\ > > > More insults. Answer me this: if there were echo cancellation in use, why > would > DTMF detection performance depend on the far-end handset's match to the > loop? > > And the follow-up question (which you've already pretty much asked) - if > the > card doesn't echo cancel for E1s, why would it for T1s? > > As an aside, I'm not convinced that the document's not talking about return > loss > on the T1 line itself, the implication being that the T1 is being carried > on a single > pair, which makes the first sentence about E1s make a bit more sense. But > that's > just a guess. > > Aculab didn't even offer echo cancellation on Prosody for years and, > when they did, it > consumed prodigious amounts of DSP. Nonetheless, the DTMF detection > worked > perfectly well, even across 120 channels per 40MHz SHARC - there's > just no way > that those DSPs had enough horsepower to do echo cancellation across > that manychannels. > > > This page http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html > seems to support what you say. It also implies DTMF detection sucks > unless you echo cancel. The statement "If the outgoing signal is a tone > of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" > is telling you to keep your outgoing signal in the same frequency range > as dial-tone where the dial-tone filter on the DTMF receiver will > obviate the need for an echo canceller. They are freely admitting > exactly what I have said. If you want a normal IVR with cut-through to > work you better turn that echo canceller on. > > My only experience with Aculab was fitting a box designed by other > people into a system. That one definitely echo cancelled, as it worked > as well as the Dialogic based boxes we developed ourselves. > > > That only holds true if your premise - that you need echo cancellation for > good > DTMF detection - is correct, which I don't believe it is. > > An Asterisk box with an el-cheapo quad E1 card in that I use for > TDM-SIP gatewaying > detects DTMF perfectly well with no echo cancellation. > > > You must have very low standards for "works well". > > > Nothing like a good old ad hominem attack. Beats reasoned argument any > day. > > You just don't need echo cancellation to achieve perfectly acceptable > DTMF detection. > > > Well, not if you expect people to wait for silence before entering DTMF, > but who would tolerate that these days? Cut through has been de rigeur > since the late 80s. > > > Oh, for pity's sake, you get perfectly good cut through without echo > cancellation. > Humour me and draw a quick mental picture of the spectrum of a random bit > of > speech at -20dBm; now add tones at -10dBm and -7dBm. They stick out like > a pair of sore thumbs. > > I'm sure it's quite possible to come up with a pathological case - e.g. > cut-through > against a 1kHz milliwatt tone, but that sort of thing just doesn't happen > in real- > life IVR applications. > > ASR - yes, maybe, but surely only in the case where the application > requires barge-in; > even then, I'd be interested to see some test results, particuarly > where the outbound prompt > is killed the moment the ASR reports start of speech. > > > Doesn't any sane system expect barge in to be nearly as reliable as > waiting for silence? Who would tolerate something that doesn't? It has > been a standard expectation of any decent IVR since they began. > > > Sorry - ASR with barge-in has been a standard expectation since the first > IVRs? > > I'm afraid that your original bald claim - that "IVRs badly need echo > cancellation" is simply > wrong, misleading and irresponsible: those believing it will end up > spending large sums > of money on technology which they probably do not need. > > > You must have very low standards for what works well. If you suggest > people leave out echo cancellation you are just asking for customer > service issues down the line. That whole Aculab page is a clear response > to just such issues they had, which forced them to add the necessary > improvements. > > > Repeating you ad-hominem really doesn't make it any stronger, I'm afraid. > And > the Aculab page you refer to offers four solutions for problems caused by > far- > end echo, of which cancellation is just one; not playing a stationary tone > above 600Hz > is another. > > Do you have any real-world samples of DTMF+echo which give your DTMF > detection code trouble? > > --Dave > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Arnaldo M Pereira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090317/153c92ec/attachment-0002.html From mattdfong at gmail.com Tue Mar 17 21:39:36 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 18 Mar 2009 11:39:36 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> Message-ID: <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> Hi Anthony, thanks for the reply. I've searched thru jira, and didn't find anything when searching for fifo that was recently updated or related, except http://jira.freeswitch.org/browse/MODAPP-189 and I'm not sure if this does what I need. Was this what you were referring to? Thanks. --matt 2009/3/17 Anthony Minessale > there is a patch in jira that will implement this feature about to be added > > > > 2009/3/17 Matthew Fong > >> I apologize if this is a double post to -dev. I'm not sure why I don't see >> my message appearing, so I'm going to try again in the -user list (first >> timer posting here ;). >> >> I have a situation where it would be useful for a caller not to be hungup, >> after finishing the "fifo in" execution (when the agent disconnects the call >> or the agent hangs-up). The caller is automatically hungup, in this >> situation. It would be preferable if the caller channel went further along >> the dial plan. I thought I might get lucky implementing this setting with >> hangup_after_bridge to false, but fifo does not utilize this variable. >> I tried looking thru the mod_fifo.c source, but my c skills are minimal. I >> also tried executing fifo in a lua app and setting setAutoHangup(false), but >> that also did not work. Any chance this could be done as a feature >> enhancement? Thanks. >> >> --matt >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/91727d52/attachment-0002.html From mszlazak at aol.com Tue Mar 17 23:59:54 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 02:59:54 -0400 Subject: [Freeswitch-users] Is mod vmd working? Message-ID: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> I followed these instructions for Mod_vmd except for a Windows box: http://wiki.freeswitch.org/wiki/Mod_vmd I tried testing to see if it's working by dialing the following extension: ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ???? ??? ??? ??? ??? ??? ??? However, I didn't see channel variable "vmd_detect" in the FreeSwitch console. ?? Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/a83f0caf/attachment-0002.html From sicfslist at gmail.com Wed Mar 18 00:11:49 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 18 Mar 2009 02:11:49 -0500 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> Message-ID: <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> Mark, It does work ... but I can't really attest to how well ... especially compared to other things out there. I started capturing this in CDR's to see and it didn't seem like it worked very well. If this is really critical to you, you might want to ping Ken Rice. I know he might have a better option. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/ac6546fd/attachment-0002.html From codecomplete at free.fr Tue Mar 17 02:20:50 2009 From: codecomplete at free.fr (Gilles) Date: Tue, 17 Mar 2009 10:20:50 +0100 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? Message-ID: <7.0.1.0.2.20090317101702.02701c88@fredshack.com> Hello For single-host settings, getting customers to buy a separate server just to run Freeswitch is overkill, so I'm thinking about selling just the IVR application to run on Windows. Unless a PCI card is available, the FXO connection will be provided by Sangoma's USB device. I'd like some feedback on running Freeswitch on XP and Vista: Is it ready for production use? Does it require beefy hardware? Thank you for any hint. From kawarod at laposte.net Wed Mar 18 01:27:56 2009 From: kawarod at laposte.net (rod) Date: Wed, 18 Mar 2009 12:27:56 +0400 Subject: [Freeswitch-users] Mod_limit stuck when hitting limit value In-Reply-To: <85B32876-7E49-4DDF-B92F-353DA9599DE9@avgs.ca> References: <49B91834.6050004@laposte.net> <8594AF27-C335-4746-920E-5217CBF4D928@avgs.ca> <49B93321.5080500@gmail.com> <49BA1EA5.4050201@laposte.net> <0CD7D8E5-C6E7-4BC7-822E-4CF635BCF41A@jerris.com> <49BA6608.70208@gmail.com> <49BA844C.3010409@freeswitch.org> <49BF4BF5.4080608@laposte.net> <85B32876-7E49-4DDF-B92F-353DA9599DE9@avgs.ca> Message-ID: <49C0B08C.4050008@laposte.net> thanks Mathieu. I setup an IRC account to give it a try. Comme ?a je pourrais t'embeter avec mes "pbms" :p rod Mathieu Rene wrote: > limit_hash uses a faster data structure then limit but works the same > way for tne end-user. > > viens sur IRC si t'as des questions en francais =) > > Math > > On 17-Mar-09, at 3:06 AM, rod wrote: > > >> Hi, >> >> not too hard :p >> but it's just a bad habit when I write in my native language >> (french). I >> guess that this spelling is not too common for english speaker. >> >> I'll do my best next time to write it correctly. >> >> @tamas >> you are right, we could use limit_hash the same way as limit when not >> specifying the /rate >> >> @Mathieu >> did you suggest limit_hash is more scalable than limit? But I don't >> understand why limit_hash is not suitable for data replication (DB >> lookup for limit and memory for limit_hash??), even if I don't know >> how >> to do it with limit. >> >> regards. >> >> Raymond Chandler wrote: >> >>> Tamas wrote: >>> >>> >>>> My guess is: pbm = problem :) >>>> >>>> >>>> >>> sure, but is it really that hard to spell all the way out? >>> >>> -Ray >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From oseslija at gmail.com Wed Mar 18 01:55:15 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Wed, 18 Mar 2009 09:55:15 +0100 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C03067.7070406@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> Message-ID: <4468a6770903180155r2fe4e1e1l91facb3085860982@mail.gmail.com> To share my experience: I had issues with echo with many E1 trunks in Serbia, especially when voice in between telco's network went to well known bad analog lines. I used OSLEC and I was fortunate to have Steve to complain to, he helped patching it further after my beta testing. Not many people would do that imho. I now switched to Sangoma cards with Octasic chips and occasionally would still hear certain echo. My view is that here some echo cancelling solution is very necessary, otherwise whole VoIP business comes up to bad reputation People would just not listen to themselves speaking, even using $400 phone. Regards, Ognjen 2009/3/18 David Knell > Steve Underwood wrote: > > [whopping big snip] > > > The first bit of that's a tad patronising, isn't it, > > > You are the one who started out being offensive. > > > I'm sorry if you find disagreement offensive; you might not wish to read > beyond this > point if so. > > and, in the case of the decade-old Aculab > cards which which I'm most familiar, is also untrue. > > > I can't find too much about the old cards on the web now, but I found http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html > which is pretty much a copy and paste from the old Dialogic web pages, > and you'll see it says "Cut through : Local echo cancellation permits > 100% detection with a >4.5 dB return loss line". The Aculabs did the > same thing for sure. They just couldn't work without cancellation. There > were some very early Dialogic cards, using DTMF receiver chips and OKI > ADPCM chips, and had no general purpose DSPs. They performed really > badly because of the lack of cancellation, and were quickly replaced > with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms > into a Motorola 56k DSP chips. > > > The same document, under the bit which you've quoted, says: > "(E-1) Digital trunks use separate transmit and receive paths to network. > Performance dependent on far end handset's match to local analog loop." > - i.e. the card does no echo cancellation. > > Aculab didn't even offer echo cancellation on Prosody for years and, when > they did, it > consumed prodigious amounts of DSP. Nonetheless, the DTMF detection worked > perfectly well, even across 120 channels per 40MHz SHARC - there's just no > way > that those DSPs had enough horsepower to do echo cancellation across that > many > channels. > > An Asterisk box with an el-cheapo quad E1 card in that I use for TDM-SIP > gatewaying > detects DTMF perfectly well with no echo cancellation. > > You just don't need echo cancellation to achieve perfectly acceptable DTMF > detection. > > ASR - yes, maybe, but surely only in the case where the application > requires barge-in; > even then, I'd be interested to see some test results, particuarly where > the outbound prompt > is killed the moment the ASR reports start of speech. > > I'm afraid that your original bald claim - that "IVRs badly need echo > cancellation" is simply > wrong, misleading and irresponsible: those believing it will end up > spending large sums > of money on technology which they probably do not need. > > --Dave > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/553ae739/attachment-0002.html From helmut.kuper at ewetel.de Wed Mar 18 02:20:09 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 18 Mar 2009 10:20:09 +0100 Subject: [Freeswitch-users] openZAP disconnect cause wrong? Message-ID: <49C0BCC9.4000509@ewetel.de> Hello, I'm not sure whether the following is a bug or a config issue: I found this in my log file: 2009-03-18 10:07:00 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: USER_BUSY 2009-03-18 10:07:00 [DEBUG] mod_dptools.c:2025 audio_bridge_function() Continue on fail [true]: Cause: USER_BUSY 2009-03-18 10:07:00 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:5/2850 [CS_EXECUTE] [NORMAL_CLEARING] FS obviously doesn't pass through the disconnect cause from Bridge app to openzap module. Analyzing the corresponding q931.pcap trace confirms this. Do I have to configure it somewhere e.g. a mapping or so, or is this a bug? regrads helmut From helmut.kuper at ewetel.de Wed Mar 18 02:40:22 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 18 Mar 2009 10:40:22 +0100 Subject: [Freeswitch-users] openZAP disconnect cause wrong? In-Reply-To: <49C0BCC9.4000509@ewetel.de> References: <49C0BCC9.4000509@ewetel.de> Message-ID: <49C0C186.3070307@ewetel.de> Hello, ok found it ... was a configuration issue due to the "continue on fail = true" variable in my dialplan. "Hangup" application fixed this :) Sorry for the post. regards helmut On 18.03.2009 10:20, Helmut Kuper wrote: > Hello, > > I'm not sure whether the following is a bug or a config issue: > > From andy at fabulous4.co.uk Wed Mar 18 04:20:11 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Wed, 18 Mar 2009 11:20:11 -0000 Subject: [Freeswitch-users] Losing Gateway registration Message-ID: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/6df3c513/attachment-0002.html From mszlazak at aol.com Wed Mar 18 05:04:37 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 08:04:37 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> Message-ID: <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> SDR? I'm wondering why there was nothing in the console showing the channel variable ${vmd_detect} as the wiki says there should be: Mark -----Original Message----- From: Shelby Ramsey To: freeswitch-users at lists.freeswitch.org Sent: Wed, 18 Mar 2009 12:11 am Subject: Re: [Freeswitch-users] Is mod vmd working? Mark, It does work ... but I can't really attest to how well ... especially compared to other things out there.? I started capturing this in CDR's to see and it didn't seem like it worked very well. If this is really critical to you, you might want to ping Ken Rice.? I know he might have a better option. SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/17cb612a/attachment-0002.html From dave at 3c.co.uk Wed Mar 18 06:00:12 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 18 Mar 2009 07:00:12 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> Message-ID: <49C0F05C.5090204@3c.co.uk> Hi Arnaldo, That's interesting - Brasil was my first proper IVR installation: one with Embratel in Sao Paulo, and then a couple with TeleRJ. I remember landing at Sao Paulo airport for the first time at 7 a.m. with instructions to "meet a fat man called Ferrari" unsure as to whether I was in some sort of elaborate hoax (I wasn't, and he was), and learning my first three words of Portuguese as we left the car park: filho da puta, of course. Those had no EC. DTMF detection worked fine, and the audio quality of the IVR recordings was perfect, which is what you'd expect: EC doesn't alter the IVR->caller audio at all. A TDM->SIP->TDM type application is a different animal: you've got the added latency of packetisation/jitter buffering/etc. which pretty much makes echo cancellation a must. --Dave > Sharing my humble experience: in Brazil we usually need echo > cancellation to have reliable DTMF detection _and_ voice quality over > E1 lines (be it on MFC/R2 - r2d - or ISDN PRI lines), either for > sip/tdm gateway devices or IVR applications. > > Usually there's no need for echo cancellation on links from some > Telcos, in some specific places. But we need it in the majority of > cases, even when my box is just a gateway between legacy pbxes. > > This represents just a subset of the available E1s in the world and > it's just a practical experience, but it's a fact for me. If I don't > have a card with echo cancellation, I don't offer reliability to my > customer; I've done that in the past and didn't work out. > > I'm not theoretically discussing anything, just sharing what I've been > through in the last 4 or 5 years. From sicfslist at gmail.com Wed Mar 18 06:07:55 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Wed, 18 Mar 2009 08:07:55 -0500 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> Message-ID: <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> Mark, Because it didn't detect a "beep". It will be be there as vmd_detect=true if it does. I'm not sure exactly how reliable it's "beep" detection is. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/36bc0a0c/attachment-0002.html From steveu at coppice.org Wed Mar 18 06:12:10 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 21:12:10 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C05641.7070309@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> Message-ID: <49C0F32A.90802@coppice.org> OK, one last go and I give up. Lets look at the documentation for Dialogic springware. This is the DSP package that loads in their cards or runs on the host in HMP applications. It does things like DTMF generation and detection for all Dialogic cards except the DM3 series. The documentation says: *PerfectDigit DTMF Signaling* ? DSP-based DTMF (touchtone) detection algorithm optimized for lowest talk-off and play-off susceptibility in the industry. The system will not easily be fooled by mistaking human speech for DTMF tones. ? Minimum tone duration and interdigit delay times accurately handle speed dialing typical of "power users" ? Utilizes echo cancellation which results in superior cut through for accurate DTMF tone interpretation during voice file playback within a broad range of network/switch environments ? DTMF outbound dialing generated by DSP for accuracy and flexibility (dialing levels are adjustable to meet a variety of global PTT requirements) Detecting supervisory tones on phone lines is trivial. Not falsely detecting them is where things get interesting. The standard test for DTMF receivers is a set of cassette tapes from Bellcore containing about 3 hours of snippets from real telephone calls in North America. Most DTMF receiver chips get a few hundred false DTMF hits in those 3 hours. Dialogic get 20 something. My DTMF receiver gets 19. The reason its hard to detect these things reliably is voice doesn't sit there nicely at one level. Its level and its spectrum bounce all over the place, and a real DTMF digit is only there for 40ms or so. I defy anyone to visually identify a 40ms DTMF digit amongst real dynamic speech if it isn't *way* above the voice in amplitude. This is why your phone has to mute your voice when you press a digit. The DTMF receiver has no chance of reliable detection with speech and digits mixed. In the few special cases where concurrent speech and signaling tone are present on the PSTN (e.g. 2280Hz signaling in .eu and 2600Hz in .us) the signaling sequence is very carefully constructed to avoid confusing the system. DTMF is never used in that way. There is one obvious special case where all DTMF receivers need to tolerate spillback. They need to differentiate between dialing tone and DTMF on the first digit you dial. They do this very simply. Dialing tone was chosen to be pretty low frequencies - 350Hz + 440Hz, 425Hz + 475Hz and similar pairings. The lowest DTMF tone is well above this. An aggressive low pass filter in the DTMF receiver removes the dial tone spillback, while barely affecting the lowest DTMF tone. This was the original design of DTMF, but...... IVRs changed all that. Their DTMF receivers are expected to work amidst outgoing prompts, which may be going to phones with an awful match to the line. The spillback can be huge. The good IVR hardware suppliers, like Dialogic, very quickly added echo cancellation to their cards. I can say a lot of negative things about Dialogic, but one thing they did really well was their DTMF cut-through. When people get used to an IVR they expect to hammer in digit sequences as fast as they can, in the face of a machine desperately trying to play voice prompts to them. Dialogic cards do this really well, on lines of all types, and on networks of varying quality. This would be impossible without echo cancellation. David Knell wrote: > Steve Underwood wrote: >> David Knell wrote: >> >>> Steve Underwood wrote: >>> >>>> [whopping big snip] >>>> >>>> >>>>> The first bit of that's a tad patronising, isn't it, >>>>> >>>>> >>>> You are the one who started out being offensive. >>>> >>>> >>> I'm sorry if you find disagreement offensive; you might not wish to >>> read beyond this >>> point if so. >>> >>>>> and, in the case of the decade-old Aculab >>>>> cards which which I'm most familiar, is also untrue. >>>>> >>>>> >>>> I can't find too much about the old cards on the web now, but I found >>>> http://www.amdevcomm.com/voice-mail-products/voice-mail-components/dialogic/dti_sc.html >>>> which is pretty much a copy and paste from the old Dialogic web pages, >>>> and you'll see it says "Cut through : Local echo cancellation permits >>>> 100% detection with a >4.5 dB return loss line". The Aculabs did the >>>> same thing for sure. They just couldn't work without cancellation. There >>>> were some very early Dialogic cards, using DTMF receiver chips and OKI >>>> ADPCM chips, and had no general purpose DSPs. They performed really >>>> badly because of the lack of cancellation, and were quickly replaced >>>> with cards that put the OKI ADPCM, DTMF anf echo cancellation algorithms >>>> into a Motorola 56k DSP chips. >>>> >>>> >>> The same document, under the bit which you've quoted, says: >>> "(E-1) Digital trunks use separate transmit and receive paths to network. >>> Performance dependent on far end handset's match to local analog loop." >>> - i.e. the card does no echo cancellation. >>> >> Your messages are starting to looked deranged. Why would they only apply >> echo cancellation to T1s? Its a bizarre idea, and you must realise its >> wrong. Are you so desperate to support a wrong answer you'll clutch at >> straws? :-\ >> > More insults. Answer me this: if there were echo cancellation in use, > why would > DTMF detection performance depend on the far-end handset's match to > the loop? > > And the follow-up question (which you've already pretty much asked) - > if the > card doesn't echo cancel for E1s, why would it for T1s? > > As an aside, I'm not convinced that the document's not talking about > return loss > on the T1 line itself, the implication being that the T1 is being > carried on a single > pair, which makes the first sentence about E1s make a bit more sense. > But that's > just a guess. >>> Aculab didn't even offer echo cancellation on Prosody for years and, >>> when they did, it >>> consumed prodigious amounts of DSP. Nonetheless, the DTMF detection >>> worked >>> perfectly well, even across 120 channels per 40MHz SHARC - there's >>> just no way >>> that those DSPs had enough horsepower to do echo cancellation across >>> that manychannels. >>> >> This page >> http://www.aculab.com/support/pdf_documents/v6_solaris/ting/pubdoc/an-dtmf-det-issues.html >> seems to support what you say. It also implies DTMF detection sucks >> unless you echo cancel. The statement "If the outgoing signal is a tone >> of some sort (e.g. a 'beep'), ensure that its frequency is below 600Hz" >> is telling you to keep your outgoing signal in the same frequency range >> as dial-tone where the dial-tone filter on the DTMF receiver will >> obviate the need for an echo canceller. They are freely admitting >> exactly what I have said. If you want a normal IVR with cut-through to >> work you better turn that echo canceller on. >> >> My only experience with Aculab was fitting a box designed by other >> people into a system. That one definitely echo cancelled, as it worked >> as well as the Dialogic based boxes we developed ourselves. >> > That only holds true if your premise - that you need echo cancellation > for good > DTMF detection - is correct, which I don't believe it is. >>> An Asterisk box with an el-cheapo quad E1 card in that I use for >>> TDM-SIP gatewaying >>> detects DTMF perfectly well with no echo cancellation. >>> >> You must have very low standards for "works well". >> > Nothing like a good old ad hominem attack. Beats reasoned argument any > day. >>> You just don't need echo cancellation to achieve perfectly acceptable >>> DTMF detection. >>> >> Well, not if you expect people to wait for silence before entering DTMF, >> but who would tolerate that these days? Cut through has been de rigeur >> since the late 80s. >> > Oh, for pity's sake, you get perfectly good cut through without echo > cancellation. > Humour me and draw a quick mental picture of the spectrum of a random > bit of > speech at -20dBm; now add tones at -10dBm and -7dBm. They stick out like > a pair of sore thumbs. > > I'm sure it's quite possible to come up with a pathological case - > e.g. cut-through > against a 1kHz milliwatt tone, but that sort of thing just doesn't > happen in real- > life IVR applications. >>> ASR - yes, maybe, but surely only in the case where the application >>> requires barge-in; >>> even then, I'd be interested to see some test results, particuarly >>> where the outbound prompt >>> is killed the moment the ASR reports start of speech. >>> >> Doesn't any sane system expect barge in to be nearly as reliable as >> waiting for silence? Who would tolerate something that doesn't? It has >> been a standard expectation of any decent IVR since they began. >> > Sorry - ASR with barge-in has been a standard expectation since the > first IVRs? >>> I'm afraid that your original bald claim - that "IVRs badly need echo >>> cancellation" is simply >>> wrong, misleading and irresponsible: those believing it will end up >>> spending large sums >>> of money on technology which they probably do not need. >>> >> You must have very low standards for what works well. If you suggest >> people leave out echo cancellation you are just asking for customer >> service issues down the line. That whole Aculab page is a clear response >> to just such issues they had, which forced them to add the necessary >> improvements. >> > Repeating you ad-hominem really doesn't make it any stronger, I'm > afraid. And > the Aculab page you refer to offers four solutions for problems caused > by far- > end echo, of which cancellation is just one; not playing a stationary > tone above 600Hz > is another. Doesn't "don't use frequencies above 600Hz" mean they won't work very well with voice present? Their "solutions" are use echo cancellation or don't create cut-through situations. That whole Aculab page is skirting around making a direct statement that reliable cut-through demands echo cancellation. > > Do you have any real-world samples of DTMF+echo which give your DTMF > detection code trouble? Any analogue line with substantial spillback. Huge numbers of lines have spillback gain only a few dB below the true receive gain. You don't normally notice this, as with a short delay its just a bit of pleasant reverb. Its becomes a problem with high latency VoIP paths stretch that delay. Its also a problem for any equipment which needs a clean signal from the far end. A modem for example, or an IVR. Echo cancellation is the only practical solution. Regards, Steve From nik.middleton at noblesolutions.co.uk Wed Mar 18 06:16:46 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Mar 2009 13:16:46 -0000 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> Message-ID: Another issue with this module is the resources it consumes. We had it running on 50 calls yesterday and the cpu's all went to 90+% Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shelby Ramsey Sent: 18 March 2009 13:08 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Is mod vmd working? Mark, Because it didn't detect a "beep". It will be be there as vmd_detect=true if it does. I'm not sure exactly how reliable it's "beep" detection is. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/4272f8e3/attachment-0002.html From steveu at coppice.org Wed Mar 18 06:25:44 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Mar 2009 21:25:44 +0800 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> Message-ID: <49C0F658.4020006@coppice.org> Nik Middleton wrote: > > Another issue with this module is the resources it consumes. We had it > running on 50 calls yesterday and the cpu?s all went to 90+% > That's odd. Something must be fouling up, as the algorithm he used should be very lightweight. Steve From intralanman at freeswitch.org Wed Mar 18 06:40:58 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 18 Mar 2009 09:40:58 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C0F05C.5090204@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> Message-ID: <49C0F9EA.3000200@freeswitch.org> David Knell wrote: > Hi Arnaldo, > > That's interesting - Brasil was my first proper IVR installation: one > with Embratel in Sao Paulo, and then a couple with TeleRJ. I remember > landing at Sao Paulo airport for the first time at 7 a.m. with > instructions to "meet a fat man called Ferrari" unsure as to whether I > was in some sort of elaborate hoax (I wasn't, and he was), and learning > my first three words of Portuguese as we left the car park: filho da > puta, of course. > What's interesting to me is.... everyone on this thread except you has said that in real-world scenarios, they need the EC for reliability. One of which, does signal processing programming professionally. It seems to me that if you "build a better mouse trap" you must know what's involved in making it work properly. I'm not sure what your background really is, but you'd be hard pressed to match up to Steve's reputation and/or experience. That said, it might be a good idea to just agree to disagree as this is starting to sound like the faxing over IP talks I hear a lot. (i.e. "faxing over g.711u with no t.38 works fine for me") Where it might work for some people by some mysterious phenomena, it won't work for the general public. And telling people that they don't need EC, where so many have already said that they obviously do, is just as irresponsible, IMHO, as you claiming Steve was for telling them that they don't need it. -Ray From brian at freeswitch.org Wed Mar 18 07:07:55 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Mar 2009 09:07:55 -0500 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: References: Message-ID: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: > Hi, > > I've recently ugrade to version 1.02 of freeswitch and am having > some problems with my gateway registrations. The gateway > successfully registers with my voip provider when freeswitch first > starts but if left running it seems to loose it's connection to my > voip provider. I can get it to reconnect with a sofia restart. I'm > using the same provider and user account as with the old version of > the software. Can you suggest any reaosn why this may be happening > and how I can prevent it? > > Many thanks > Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/ec96a7e9/attachment-0002.html From mrene_lists at avgs.ca Wed Mar 18 07:45:39 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 18 Mar 2009 10:45:39 -0400 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> References: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> Message-ID: <2365564C-92A8-483E-9FCB-34EBE71EC256@avgs.ca> if you are behind NAT it is possible that your router "forgot" the mapping betweeen FS and your provider, try adding to your gateway. Math On 18-Mar-09, at 10:07 AM, Brian West wrote: > Upgrade to 1.03 or SVN Trunk > > /b > > On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: > >> Hi, >> >> I've recently ugrade to version 1.02 of freeswitch and am having >> some problems with my gateway registrations. The gateway >> successfully registers with my voip provider when freeswitch first >> starts but if left running it seems to loose it's connection to my >> voip provider. I can get it to reconnect with a sofia restart. I'm >> using the same provider and user account as with the old version of >> the software. Can you suggest any reaosn why this may be happening >> and how I can prevent it? >> >> Many thanks >> Andy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/ae829584/attachment-0002.html From mszlazak at aol.com Wed Mar 18 08:32:32 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 11:32:32 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> Message-ID: <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> I added a voicemail tag in 5555 to a default extension 1001, I hear the voicemail beep but still don't see vmd_detect. Mark -----Original Message----- From: Shelby Ramsey To: freeswitch-users at lists.freeswitch.org Sent: Wed, 18 Mar 2009 6:07 am Subject: Re: [Freeswitch-users] Is mod vmd working? Mark, Because it didn't detect a "beep".? It will be be there as vmd_detect=true if it does.? I'm not sure exactly how reliable it's "beep" detection is.? SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/b34a1129/attachment-0002.html From mike at jerris.com Wed Mar 18 08:33:28 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Mar 2009 11:33:28 -0400 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: <7.0.1.0.2.20090317101702.02701c88@fredshack.com> References: <7.0.1.0.2.20090317101702.02701c88@fredshack.com> Message-ID: There is currently no openzap (sangoma, etc) support on windows, we hope this will be coming soon. Mike On Mar 17, 2009, at 5:20 AM, Gilles wrote: > Hello > > For single-host settings, getting customers to buy a separate server > just to run Freeswitch is overkill, so I'm thinking about selling > just the IVR application to run on Windows. Unless a PCI card is > available, the FXO connection will be provided by Sangoma's USB > device. > > I'd like some feedback on running Freeswitch on XP and Vista: Is it > ready for production use? Does it require beefy hardware? > > Thank you for any hint. From mike at jerris.com Wed Mar 18 08:34:24 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 18 Mar 2009 11:34:24 -0400 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: <20090318023154.GA16523@jdc.jasonjgw.net> References: <20090318014659.GA15840@jdc.jasonjgw.net> <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> <20090318023154.GA16523@jdc.jasonjgw.net> Message-ID: On Mar 17, 2009, at 10:31 PM, Jason White wrote: > Brian West wrote: >> if you installed the ssl devel stuff AFTER you configured you'll need >> to reconfigure. > > I'm reasonably sure it was installed already, unless it was pulled > in recently > by a package upgrade. > > The configure script needs to look in /usr/include/openssl for the > headers. > I'll have a look at config.log and try to work out what it looked > for and why > it didn't find it. you will have to look in the config.log in libs/sofia-sip Mike From Mark.Tabron at rnid-typetalk.org.uk Wed Mar 18 08:52:14 2009 From: Mark.Tabron at rnid-typetalk.org.uk (MarkTab) Date: Wed, 18 Mar 2009 08:52:14 -0700 (PDT) Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> Message-ID: <22582281.post@talk.nabble.com> We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things! Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819 http://pastebin.freeswitch.org/7819 I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818 http://pastebin.freeswitch.org/7818 Getting there (slowly) Mark. mercutioviz wrote: > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: >> Another update - this time (part) good news! Decided to run wancfg_tdmapi >> again, using the same settings as we always did, and we can now make >> external calls. I suspect that whatever BT did yesterday kicked the >> circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > >> >> However placing an external call into FS isn't as successful, looks like >> it can't assign a channel and terminates the call. >> > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Problem-dialing-out-via-E1-tp22479047p22582281.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Mark.Tabron at rnid-typetalk.org.uk Wed Mar 18 08:55:50 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Wed, 18 Mar 2009 15:55:50 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> Message-ID: <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things! Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819 I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818 Getting there (slowly) Mark. On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron wrote: > Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed." > > However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. > Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 March 2009 15:48 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron wrote: > Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed." > > However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. > Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- From mattdfong at gmail.com Wed Mar 18 08:59:07 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 18 Mar 2009 22:59:07 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> Message-ID: <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> I upgraded to FreeSWITCH Version 1.0.trunk (12654M) but caller is still being hungup (and not continuing on with dialplan) after agent disconnect with hangup_after_bridge=false Is there a separate patch I need to apply? Thanks. --matt On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: > Hi Anthony, thanks for the reply. > I've searched thru jira, and didn't find anything when searching for fifo > that was recently updated or related, except > > http://jira.freeswitch.org/browse/MODAPP-189 > > and I'm not sure if this does what I need. Was this what you were referring > to? Thanks. > > --matt > > 2009/3/17 Anthony Minessale > > there is a patch in jira that will implement this feature about to be added >> >> >> >> 2009/3/17 Matthew Fong >> >>> I apologize if this is a double post to -dev. I'm not sure why I don't >>> see my message appearing, so I'm going to try again in the -user list (first >>> timer posting here ;). >>> >>> I have a situation where it would be useful for a caller not to be >>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>> the call or the agent hangs-up). The caller is automatically hungup, in this >>> situation. It would be preferable if the caller channel went further along >>> the dial plan. I thought I might get lucky implementing this setting with >>> hangup_after_bridge to false, but fifo does not utilize this variable. >>> I tried looking thru the mod_fifo.c source, but my c skills are minimal. >>> I also tried executing fifo in a lua app and setting setAutoHangup(false), >>> but that also did not work. Any chance this could be done as a feature >>> enhancement? Thanks. >>> >>> --matt >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/9bec90cd/attachment-0002.html From Prometheus001 at gmx.net Wed Mar 18 10:18:16 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 18 Mar 2009 18:18:16 +0100 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <49C12CD8.7020203@gmx.net> 2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch there was a timer problem which was not solved yet. This caused channels to be busy in my case. I am not sure whether this is solved yet. Can anybody confirm? Best regards Peter Mark Tabron schrieb: > We're a couple more steps forward from yesterday. Turned out some of my > regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has > an extra space before one of the closing brackets in the default.xml > example. After staring at the screen all day it's funny how you miss > these things! > > Situation now is I can get the call into FS but, it rings the extension > for a fraction of a second then the call drops. Here's the contents of > the public and default dialplans I'm using (as per example in the wiki) > and the debug - http://pastebin.freeswitch.org/7819 > > I'm also seeing another issue when placing subsequent inbound calls, > they bounce if hitting the same channel the first call came in to > (typically /1:1). Again, grabbed a debug of this - > http://pastebin.freeswitch.org/7818 > > Getting there (slowly) > > Mark. > > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: > >> Another update - this time (part) good news! Decided to run >> > wancfg_tdmapi again, using the same settings as we always did, and we > can now make external calls. I suspect that whatever BT did yesterday > kicked the circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > > >> However placing an external call into FS isn't as successful, looks >> > like it can't assign a channel and terminates the call. > > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 17 March 2009 15:48 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: > >> Another update - this time (part) good news! Decided to run >> > wancfg_tdmapi again, using the same settings as we always did, and we > can now make external calls. I suspect that whatever BT did yesterday > kicked the circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > > >> However placing an external call into FS isn't as successful, looks >> > like it can't assign a channel and terminates the call. > > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > -------------------------------------------------------------------------------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > -------------------------------------------------------------------------------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Wed Mar 18 10:24:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Mar 2009 10:24:01 -0700 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> Message-ID: <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> 2009/3/18 : > I added a voicemail tag in 5555 to a default extension 1001, I hear the > voicemail beep but still don't see vmd_detect. > > Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering machine beeps and vm beeps on cell phone voice mails. -MC From msc at freeswitch.org Wed Mar 18 10:27:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Mar 2009 10:27:10 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <49C12CD8.7020203@gmx.net> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> <49C12CD8.7020203@gmx.net> Message-ID: <87f2f3b90903181027y5937f48bxa5aecea367db4cc0@mail.gmail.com> On Wed, Mar 18, 2009 at 10:18 AM, Peter P GMX wrote: > 2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch > there was a timer problem which was not solved yet. This caused channels > to be busy in my case. > > I am not sure whether this is solved yet. Can anybody confirm? > We're using ozmod_libpri which has it's own PRI handling. So far, so good... -MC From anthony.minessale at gmail.com Wed Mar 18 12:00:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Mar 2009 14:00:14 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> Message-ID: <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> This is the patch http://jira.freeswitch.org/browse/MODAPP-237 it's not added yet. 2009/3/18 Matthew Fong > I upgraded to > FreeSWITCH Version 1.0.trunk (12654M) > > but caller is still being hungup (and not continuing on with dialplan) > after agent disconnect with hangup_after_bridge=false > > Is there a separate patch I need to apply? Thanks. > > --matt > > > On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: > >> Hi Anthony, thanks for the reply. >> I've searched thru jira, and didn't find anything when searching for fifo >> that was recently updated or related, except >> >> http://jira.freeswitch.org/browse/MODAPP-189 >> >> and I'm not sure if this does what I need. Was this what you were >> referring to? Thanks. >> >> --matt >> >> 2009/3/17 Anthony Minessale >> >> there is a patch in jira that will implement this feature about to be >>> added >>> >>> >>> 2009/3/17 Matthew Fong >>> >>>> I apologize if this is a double post to -dev. I'm not sure why I don't >>>> see my message appearing, so I'm going to try again in the -user list (first >>>> timer posting here ;). >>>> >>>> I have a situation where it would be useful for a caller not to be >>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>> situation. It would be preferable if the caller channel went further along >>>> the dial plan. I thought I might get lucky implementing this setting with >>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>> I tried looking thru the mod_fifo.c source, but my c skills are minimal. >>>> I also tried executing fifo in a lua app and setting setAutoHangup(false), >>>> but that also did not work. Any chance this could be done as a feature >>>> enhancement? Thanks. >>>> >>>> --matt >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/0d7c7233/attachment-0002.html From mszlazak at aol.com Wed Mar 18 15:12:20 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 18:12:20 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com><35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com><8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> Message-ID: <8CB762DBDEB5479-1554-817@mblk-d43.sysops.aol.com> Hi MC, With trunk 12638M, I tried checking vmd internally and externally to my cell. No luck at all in detecting a voicemail (beep). I used the following extensions to test this, maybe they are in error. If not then how else can I detect from FS that I got voicemail in a phone agnostic way (i.e, pots & sip). ??? ??? ??? Mark. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 18 Mar 2009 10:24 am Subject: Re: [Freeswitch-users] Is mod vmd working? 2009/3/18 : > I added a voicemail tag in 5555 to a default extension 1001, I hear the > voicemail beep but still don't see vmd_detect. > > Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering machine beeps and vm beeps on cell phone voice mails. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/3c414272/attachment-0002.html From kjv at ken-ton.com Wed Mar 18 15:39:59 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Wed, 18 Mar 2009 18:39:59 -0400 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: References: <20090318014659.GA15840@jdc.jasonjgw.net> <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> <20090318023154.GA16523@jdc.jasonjgw.net> Message-ID: Was this ever resolved? If we're missing something in the documentation, I'd like to make sure it's in there. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3239 x0 On Mar 18, 2009, at 11:34 AM, Michael Jerris wrote: > > On Mar 17, 2009, at 10:31 PM, Jason White wrote: > >> Brian West wrote: >>> if you installed the ssl devel stuff AFTER you configured you'll >>> need >>> to reconfigure. >> >> I'm reasonably sure it was installed already, unless it was pulled >> in recently >> by a package upgrade. >> >> The configure script needs to look in /usr/include/openssl for the >> headers. >> I'll have a look at config.log and try to work out what it looked >> for and why >> it didn't find it. > > you will have to look in the config.log in libs/sofia-sip > > Mike > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/4a424864/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: PGP.sig Type: application/pgp-signature Size: 833 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/4a424864/attachment-0002.bin From brian at freeswitch.org Wed Mar 18 15:43:40 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Mar 2009 17:43:40 -0500 Subject: [Freeswitch-users] TLS support in Debian build In-Reply-To: References: <20090318014659.GA15840@jdc.jasonjgw.net> <795409B1-06BB-4908-B0DA-9A03C551EBDF@freeswitch.org> <20090318023154.GA16523@jdc.jasonjgw.net> Message-ID: I thought we had... hrm. /b On Mar 18, 2009, at 5:39 PM, Karl Vesterling wrote: > Was this ever resolved? > If we're missing something in the documentation, I'd like to make > sure it's in there. > > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3239 x0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/d72d9983/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Mar 18 16:22:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Mar 2009 23:22:14 -0000 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com><35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com><8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> Message-ID: Hmm, Well We're connected direct to E1's and it doesn't work reliably here. That said, DTMF detect does recognise the beeps most of the time. Perhaps there's a regional variation. I wonder if it's country specific. The code looks logical. When I get some time I'll have a look at it and see how it can be improved. The concept is great and is much better that sniffing out human voice as that's prone to false positives. Much better to assume human and machine. Nothing worse than a silent call. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 March 2009 17:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Is mod vmd working? 2009/3/18 : > I added a voicemail tag in 5555 to a default extension 1001, I hear the > voicemail beep but still don't see vmd_detect. > > Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering machine beeps and vm beeps on cell phone voice mails. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed Mar 18 17:15:34 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Mar 2009 17:15:34 -0700 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> Message-ID: <87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com> Ironically, I've used tone_detect to try and trap SIT tones and I found that answering machines in the USA seem to all send a beep in the same freq range as American SIT tones... :) -MC On Wed, Mar 18, 2009 at 4:22 PM, Nik Middleton wrote: > Hmm, > > Well We're connected direct to E1's and it doesn't work reliably here. > That said, DTMF detect does recognise the beeps most of the time. > Perhaps there's a regional variation. ?I wonder if it's country > specific. ?The code looks logical. ?When I get some time I'll have a > look at it and see how it can be improved. > > The concept is great and is much better that sniffing out human voice as > that's prone to false positives. ?Much better to assume human and > machine. ?Nothing worse than a silent call. > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 18 March 2009 17:24 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Is mod vmd working? > > 2009/3/18 ?: >> I added a voicemail tag in 5555 to a default extension 1001, I hear > the >> voicemail beep but still don't see vmd_detect. >> >> Mark > > FYI, I've used mod_vmd but only in a TDM environment on outbound calls > via a PRI. It worked very well on for detecting answering ?machine > beeps and vm beeps on cell phone voice mails. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Wed Mar 18 20:16:45 2009 From: dujinfang at gmail.com (dujinfang) Date: Thu, 19 Mar 2009 11:16:45 +0800 Subject: [Freeswitch-users] Is there a way to automatically re-login gtalk account Message-ID: Hi all, mod_dingaling in client mode works well for me, but disconnected yesterday. 2009-03-18 16:57:32 [DEBUG] libdingaling.c:1545 xmpp_connect() io error 2 7 I use dl_login profile=gmail.com, and it re-login successfully. Is their a way to auto re-login after fail? Thanks. From mszlazak at aol.com Wed Mar 18 20:20:18 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 18 Mar 2009 23:20:18 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com><35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com><8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com><87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> <87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com> Message-ID: <8CB7658C3800442-84C-1EAB@WEBMAIL-DY38.sysops.aol.com> tone_detect! sounds good. BTW, was there any errors in those extensions I posted. I modified something you posted MC. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 18 Mar 2009 5:15 pm Subject: Re: [Freeswitch-users] Is mod vmd working? Ironically, I've used tone_detect to try and trap SIT tones and I found that answering machines in the USA seem to all send a beep in the same freq range as American SIT tones... :) -MC On Wed, Mar 18, 2009 at 4:22 PM, Nik Middleton wrote: > Hmm, > > Well We're connected direct to E1's and it doesn't work reliably here. > That said, DTMF detect does recognise the beeps most of the time. > Perhaps there's a regional variation. ?I wonder if it's country > specific. ?The code looks logical. ?When I get some time I'll have a > look at it and see how it can be improved. > > The concept is great and is much better that sniffing out human voice as > that's prone to false positives. ?Much better to assume human and > machine. ?Nothing worse than a silent call. > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 18 March 2009 17:24 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Is mod vmd working? > > 2009/3/18 ?: >> I added a voicemail tag in 5555 to a default extension 1001, I hear > the >> voicemail beep but still don't see vmd_detect. >> >> Mark > > FYI, I've used mod_vmd but only in a TDM environment on outbound calls > via a PRI. It worked very well on for detecting answering ?machine > beeps and vm beeps on cell phone voice mails. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090318/657e01ac/attachment-0002.html From d at d-man.org Wed Mar 18 20:27:52 2009 From: d at d-man.org (Darren Schreiber) Date: Wed, 18 Mar 2009 20:27:52 -0700 Subject: [Freeswitch-users] Issue relating mod_nibblebill In-Reply-To: References: Message-ID: Is this issue still open? I just noticed this. The error you are receiving indicates UnixODBC is installed, but not configured properly (most likely anyway). The UnixODBC drivers are kind of a pain to setup on some systems, especially CentOS, but this article may help you get it working - http://webaj.com/how-setup-mysql-dsn-datasbase-source-centos-myodbc-and-unix odbc-command-line.htm. I strongly recommend making sure the test commands they list work before trying to get UnixODBC working within FreeSWITCH. Also, it looks like you may have failed to copy the sample mod_nibblebill XML config file to /usr/local/freeswitch/conf/autoload_configs/ . You may want to give that a try. Within that file is the name of the ODBC driver being referenced - make sure that driver exists (see link above). - Darren -----Original Message----- From: JayaPrakash [mailto:jp.manchu at gmail.com] Sent: Saturday, March 14, 2009 4:49 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Issue relating mod_nibblebill Hi All, I am newbie to Freeswitch. I installed freeswitch-1.0.3 in Debian machine. I am able to make call, check presence, retrieve CDRs. I followed the installation steps given in mod_nibblebill for rating. While, installing mysql-connector-odbc, it has thrown errors related to mysql-config file, that it does not exist. Coming to mysql, mysql-client-5 and mysql-server-5 are installed. So I installed libmyodbc which is used for the same functionality. Rest of the steps are done, as given in mod-nibblebill installation. When the freeradius server is restarted, it has given the following error. 2009-03-14 14:55:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-03-14 14:55:47 [CRIT] mod_nibblebill.c:233 load_config() Cannot connect to ODBC driver/database freeswitch (user: root / pass dev)! Will you please have a look in solving this issue ? , how the issue can be solved? Thanks & Regards, JP. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From dujinfang at gmail.com Wed Mar 18 20:30:23 2009 From: dujinfang at gmail.com (seven) Date: Thu, 19 Mar 2009 11:30:23 +0800 Subject: [Freeswitch-users] Is there a way to automatically re-login gtalk account Message-ID: <49963D01-BDB1-4142-8B1F-8F22773D950E@gmail.com> I do have auto-login enabled in jingle_profile: From pablosaro at gmail.com Wed Mar 18 21:24:46 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Thu, 19 Mar 2009 01:24:46 -0300 Subject: [Freeswitch-users] FS in Solaris Message-ID: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> Hi list, Any experience building FS in Solaris using Sun Studio? Thanks Pablo From switchserver at gmail.com Thu Mar 19 00:41:26 2009 From: switchserver at gmail.com (HarryK) Date: Thu, 19 Mar 2009 00:41:26 -0700 (PDT) Subject: [Freeswitch-users] Cepstral and RSS feeds Message-ID: <22594923.post@talk.nabble.com> I have Cepstral working. Can someone please tell me how to go about having it read RSS feeds? I can have the dialplan direct it np. But I really dont have a clue how to point it at an RSS. Any help would be great, ddint find anything in the wiki. -- View this message in context: http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22594923.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yudha2008 at gmail.com Thu Mar 19 00:53:50 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 19 Mar 2009 13:23:50 +0530 Subject: [Freeswitch-users] dynamic conference In-Reply-To: <21123756.post@talk.nabble.com> References: <21053181.post@talk.nabble.com> <49490301.8040701@lists.rupa.com> <21054080.post@talk.nabble.com> <49491847.3030403@lists.rupa.com> <21069519.post@talk.nabble.com> <494A9EE2.7050507@freeswitch.org> <21123756.post@talk.nabble.com> Message-ID: *Hi, I have seen the above mail. In that all of you tried to created dynamic conference through diaplan itself using the database to insert the uuid, caller_id_number, destination_number, etc .Can one guide me set the dynamic conference and Schema for the dynamic conference. I have tried the above dialplan with my Freeswitch 1.0.2 server but i cant able to get it. Can any one guide me for the this process. -- Warm Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/19c81283/attachment-0002.html From Mark.Tabron at rnid-typetalk.org.uk Thu Mar 19 02:08:28 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Thu, 19 Mar 2009 09:08:28 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903120950h1658b9f1k1d815c1e730f0dbd@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> <49C12CD8.7020203@gmx.net> Message-ID: <11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL> So the second issue is possibly known - really could do with a fix or a workaround for this as we plan to use E1's for all incoming traffic. Can anyone shed light on the first problem (extension rings for a fraction of a second then hangs up) I mentioned below, or is that possibly part of the same issue? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P GMX Sent: 18 March 2009 17:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch there was a timer problem which was not solved yet. This caused channels to be busy in my case. I am not sure whether this is solved yet. Can anybody confirm? Best regards Peter Mark Tabron schrieb: > We're a couple more steps forward from yesterday. Turned out some of my > regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has > an extra space before one of the closing brackets in the default.xml > example. After staring at the screen all day it's funny how you miss > these things! > > Situation now is I can get the call into FS but, it rings the extension > for a fraction of a second then the call drops. Here's the contents of > the public and default dialplans I'm using (as per example in the wiki) > and the debug - http://pastebin.freeswitch.org/7819 > > I'm also seeing another issue when placing subsequent inbound calls, > they bounce if hitting the same channel the first call came in to > (typically /1:1). Again, grabbed a debug of this - > http://pastebin.freeswitch.org/7818 > > Getting there (slowly) > > Mark. > > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: > >> Another update - this time (part) good news! Decided to run >> > wancfg_tdmapi again, using the same settings as we always did, and we > can now make external calls. I suspect that whatever BT did yesterday > kicked the circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > > >> However placing an external call into FS isn't as successful, looks >> > like it can't assign a channel and terminates the call. > > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 17 March 2009 15:48 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Problem dialing out via E1 > > On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron > wrote: > >> Another update - this time (part) good news! Decided to run >> > wancfg_tdmapi again, using the same settings as we always did, and we > can now make external calls. I suspect that whatever BT did yesterday > kicked the circuit back into life. > > Good. I can't tell you how many times I've spoken to a telco when > there's a problem and the circuit magically comes back to life. They > frequently claim, "We didn't do anything." I think that's a euphemism > for "we did a reset and prayed." > > >> However placing an external call into FS isn't as successful, looks >> > like it can't assign a channel and terminates the call. > > > Be sure that you have some routing mechanism in your public.xml file. > Do you have a whole block of DID numbers? Anyway, pastebin your > public.xml and a debug trace of an incoming call, including what phone > number the caller dialed, and we'll take a look. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Save paper - don't print this email unless you need to. > > ------------------------------------------------------------------------ -------- > NOTICE from RNID Typetalk > > This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. > If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. > Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). > > > > ------------------------------------------------------------------------ -------- > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From qulix at mail.ru Thu Mar 19 04:05:23 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Thu, 19 Mar 2009 14:05:23 +0300 Subject: [Freeswitch-users] Good time, people! Message-ID: Hello! Has anybody faced such a problem with xml_curl? 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing 1000->********** in context default 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() Oversized file detected [136089828 bytes] 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error encountered! Tcpdump log tells that transaction is allright (xml dialplan is fine and etc) But FS says its oversized =\ what could be wrong? My trunk is : FreeSWITCH Version 1.0.trunk (12573). From codecomplete at free.fr Thu Mar 19 04:54:54 2009 From: codecomplete at free.fr (Gilles) Date: Thu, 19 Mar 2009 12:54:54 +0100 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: References: Message-ID: <7.0.1.0.2.20090319125259.02475630@free.fr> Michael Jerris > There is currently no openzap (sangoma, etc) support on windows, we hope this will be coming soon. I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper too. Would you say the Windows port of Freeswitch is ready to be used commercially, or I should go for a Linux box instead? Thank you. From leon at scarlet-internet.nl Thu Mar 19 04:57:46 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Mar 2009 12:57:46 +0100 Subject: [Freeswitch-users] Proxy media hickups in audio Message-ID: Hi all, I'm still undecided yet whether I need proxy-media or not. As I understand it, the only downside of enabling proxy-media is that early- media is not possible, correct ? (Or are there other reasons why I shouldn't use proxy-media ?) When I disable proxy-media I get little hickups in the audio on the outbound leg of the call, which are not there when proxy-media is enabled. I made pcap dumps on the ethernet interface for both in- and outbound legs, for both settings (proxy-media dis- and enabled), analyzed it in wireshark (no packetloss), but when extracting the audiostreams to .au and listening to it on my laptop, I hear the same hickups. Is there some other setting that may fix the problem of the hickups ? (Is this a known problem?) I'm running fs with -hp argument, am doing no transcoding and am using ALAW codec. Is it helpful if I put the pcap and .au files somewhere ? thanks & regards, Leon From intralanman at freeswitch.org Thu Mar 19 05:25:59 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 19 Mar 2009 08:25:59 -0400 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <22594923.post@talk.nabble.com> References: <22594923.post@talk.nabble.com> Message-ID: <49C239D7.4080101@freeswitch.org> HarryK wrote: > I have Cepstral working. > > Can someone please tell me how to go about having it read RSS feeds? I can > have the dialplan direct it np. But I really dont have a clue how to point > it at an RSS. Any help would be great, ddint find anything in the wiki. > > > have you tried mod_rss? -Ray From freeswitch at gnarg.org Thu Mar 19 05:08:29 2009 From: freeswitch at gnarg.org (freeswitch at gnarg.org) Date: Thu, 19 Mar 2009 13:08:29 +0100 Subject: [Freeswitch-users] FS in Solaris In-Reply-To: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> References: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> Message-ID: <49C235BD.1040905@gnarg.org> Pablo Hernan Saro wrote: > Hi list, > > Any experience building FS in Solaris using Sun Studio? http://www.voiceworks.pl/cypromis/tag/opensolaris/ Chris From stkn at freeswitch.org Thu Mar 19 06:12:53 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Thu, 19 Mar 2009 14:12:53 +0100 Subject: [Freeswitch-users] FS in Solaris In-Reply-To: <49C235BD.1040905@gnarg.org> References: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> <49C235BD.1040905@gnarg.org> Message-ID: <200903191412.53527.stkn@freeswitch.org> Am Thursday 19 March 2009 schrieb freeswitch at gnarg.org: > Pablo Hernan Saro wrote: > > > Hi list, > > > > Any experience building FS in Solaris using Sun Studio? > > http://www.voiceworks.pl/cypromis/tag/opensolaris/ > > Chris > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > There are still some issues we're currently working on. You'll have to add "--disable-visibility" to the configure line to get a working mod_sofia and softtimer. Mod_lua is broken too and i'm still trying to find a fix for that. SFEcurses and SFEncursesw may cause FS to segfault on startup (if those are installed on your system), "--disable-core-libedit-support" will fix that, but you'll loose the line edit and history feature of the FS console. stkn -- Stefan Knoblich Web: http://stkn.techmage.de/ http://oss.axsentis.de/people/stkn/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From brian at freeswitch.org Thu Mar 19 06:14:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 08:14:52 -0500 Subject: [Freeswitch-users] Good time, people! In-Reply-To: References: Message-ID: <1A6B5E65-7091-41EC-9C67-619E17632F71@freeswitch.org> Any reason you're feeding it a 130+ meg file over XML_CURL? /b On Mar 19, 2009, at 6:05 AM, ????... wrote: > Hello! > > Has anybody faced such a problem with xml_curl? > 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing 1000->********** in context default > 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() > Oversized file detected [136089828 bytes] > 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error > encountered! > > Tcpdump log tells that transaction is allright (xml dialplan is fine > and etc) > But FS says its oversized =\ what could be wrong? > > My trunk is : > FreeSWITCH Version 1.0.trunk (12573). From brian at freeswitch.org Thu Mar 19 06:15:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 08:15:51 -0500 Subject: [Freeswitch-users] Proxy media hickups in audio In-Reply-To: References: Message-ID: You shouldn't use it. It has a special use case and I suspect yours isn't it. Are you doing anything with T.38 right now? /b On Mar 19, 2009, at 6:57 AM, Leon de Rooij wrote: > I'm still undecided yet whether I need proxy-media or not. As I > understand it, the only downside of enabling proxy-media is that > early- > media is not possible, correct ? (Or are there other reasons why I > shouldn't use proxy-media ?) From leon at scarlet-internet.nl Thu Mar 19 06:30:10 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Mar 2009 14:30:10 +0100 Subject: [Freeswitch-users] Proxy media hickups in audio In-Reply-To: References: Message-ID: <8D62A4C8-C6DB-4860-B12F-3E02C3C2EE63@scarlet-internet.nl> I'm not using T38 yet, it may be nice in the future, as long faxes over alaw just don't work properly.. And also, there are these hickups now, that I don't have with proxy- media enabled.. On Mar 19, 2009, at 2:15 PM, Brian West wrote: > You shouldn't use it. It has a special use case and I suspect yours > isn't it. Are you doing anything with T.38 right now? > > /b > > On Mar 19, 2009, at 6:57 AM, Leon de Rooij wrote: > >> I'm still undecided yet whether I need proxy-media or not. As I >> understand it, the only downside of enabling proxy-media is that >> early- >> media is not possible, correct ? (Or are there other reasons why I >> shouldn't use proxy-media ?) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pablosaro at gmail.com Thu Mar 19 06:34:58 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Thu, 19 Mar 2009 10:34:58 -0300 Subject: [Freeswitch-users] FS in Solaris In-Reply-To: <200903191412.53527.stkn@freeswitch.org> References: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> <49C235BD.1040905@gnarg.org> <200903191412.53527.stkn@freeswitch.org> Message-ID: <247f8100903190634u63d2e29bm7dd0b74abf1976b3@mail.gmail.com> Well, I guess that is something I can deal with... Actually it is for benchmarking purposes. I was discussing about performance with a colleague, who is a Sr Solaris Engineer, and he recommended me to build FS in Solaris and benchmark it. He ensures that it would be really better due to Fire Engine, the Solaris' networking stack. Thank you guys for your replies. I will look into it, following your guide lines and bothering my colleague :) I hope to get interesting results to share with the community. Regards Pablo On Thu, Mar 19, 2009 at 10:12 AM, Stefan Knoblich wrote: > Am Thursday 19 March 2009 schrieb freeswitch at gnarg.org: >> Pablo Hernan Saro wrote: >> >> > Hi list, >> > >> > Any experience building FS in Solaris using Sun Studio? >> >> http://www.voiceworks.pl/cypromis/tag/opensolaris/ >> >> Chris >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > There are still some issues we're currently working on. > You'll have to add "--disable-visibility" to the configure line to get a working > mod_sofia and softtimer. > Mod_lua is broken too and i'm still trying to find a fix for that. > > SFEcurses and SFEncursesw may cause FS to segfault on startup > (if those are installed on your system), "--disable-core-libedit-support" > will fix that, but you'll loose the line edit and history feature of the FS console. > > > stkn > > -- > Stefan Knoblich > > Web: ? http://stkn.techmage.de/ ? ? ? ?http://oss.axsentis.de/people/stkn/ > Email: stkn at freeswitch.org > IRC: ? ?#freeswitch-de @ irc.freenode.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From michal.bielicki at voiceworks.pl Thu Mar 19 06:41:59 2009 From: michal.bielicki at voiceworks.pl (Michal Bielicki) Date: Thu, 19 Mar 2009 14:41:59 +0100 Subject: [Freeswitch-users] FS in Solaris In-Reply-To: <247f8100903190634u63d2e29bm7dd0b74abf1976b3@mail.gmail.com> References: <247f8100903182124p54af2668k2efeb646b51f62d@mail.gmail.com> <49C235BD.1040905@gnarg.org> <200903191412.53527.stkn@freeswitch.org> <247f8100903190634u63d2e29bm7dd0b74abf1976b3@mail.gmail.com> Message-ID: <49C24BA7.4060802@voiceworks.pl> If you do a proper side by side test, let me know the results and we will publish them. cheers Michal Pablo Hernan Saro schrieb: > Well, I guess that is something I can deal with... Actually it is for > benchmarking purposes. I was discussing about performance with a > colleague, who is a Sr Solaris Engineer, and he recommended me to > build FS in Solaris and benchmark it. He ensures that it would be > really better due to Fire Engine, the Solaris' networking stack. > Thank you guys for your replies. I will look into it, following your > guide lines and bothering my colleague :) > I hope to get interesting results to share with the community. > Regards > > Pablo > > On Thu, Mar 19, 2009 at 10:12 AM, Stefan Knoblich wrote: > >> Am Thursday 19 March 2009 schrieb freeswitch at gnarg.org: >> >>> Pablo Hernan Saro wrote: >>> >>> >>>> Hi list, >>>> >>>> Any experience building FS in Solaris using Sun Studio? >>>> >>> http://www.voiceworks.pl/cypromis/tag/opensolaris/ >>> >>> Chris >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> There are still some issues we're currently working on. >> You'll have to add "--disable-visibility" to the configure line to get a working >> mod_sofia and softtimer. >> Mod_lua is broken too and i'm still trying to find a fix for that. >> >> SFEcurses and SFEncursesw may cause FS to segfault on startup >> (if those are installed on your system), "--disable-core-libedit-support" >> will fix that, but you'll loose the line edit and history feature of the FS console. >> >> >> stkn >> >> -- >> Stefan Knoblich >> >> Web: http://stkn.techmage.de/ http://oss.axsentis.de/people/stkn/ >> Email: stkn at freeswitch.org >> IRC: #freeswitch-de @ irc.freenode.net >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From michal.bielicki at voiceworks.pl Thu Mar 19 06:43:18 2009 From: michal.bielicki at voiceworks.pl (Michal Bielicki) Date: Thu, 19 Mar 2009 14:43:18 +0100 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: References: <7.0.1.0.2.20090317101702.02701c88@fredshack.com> Message-ID: <49C24BF6.2080207@voiceworks.pl> Michael Jerris schrieb: > There is currently no openzap (sangoma, etc) support on windows, we > hope this will be coming soon. > > Mike > > On Mar 17, 2009, at 5:20 AM, Gilles wrote: > > >> Hello >> >> For single-host settings, getting customers to buy a separate server >> just to run Freeswitch is overkill, so I'm thinking about selling >> just the IVR application to run on Windows. Unless a PCI card is >> available, the FXO connection will be provided by Sangoma's USB >> device. >> >> I'd like some feedback on running Freeswitch on XP and Vista: Is it >> ready for production use? Does it require beefy hardware? >> >> Thank you for any hint. >> > > > You could use Netborder Express with it. From asannucci at gmail.com Thu Mar 19 08:28:36 2009 From: asannucci at gmail.com (Andrea) Date: Thu, 19 Mar 2009 10:28:36 -0500 Subject: [Freeswitch-users] FreeSWITCH Italian Forum Message-ID: Hi, maybe this message can considered off-topic, but i think can be interessing for FreeSWITCH community. There is a new forum on FreeSWITCH for italian people. Please visit www.freeswitch-it.org Any suggest are welcome I hope to do my english a little bit better :) Best Regards - Andrea - From leon at scarlet-internet.nl Thu Mar 19 08:34:14 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Thu, 19 Mar 2009 16:34:14 +0100 Subject: [Freeswitch-users] Proxy media hickups in audio In-Reply-To: References: Message-ID: <2E45931E-E495-4DA5-BCA1-8FB545BA2D61@scarlet-internet.nl> Brian, I put two au files here: http://www.ldr.scarlet.nl/ua-to-fs.au http://www.ldr.scarlet.nl/fs-to-mgw.au It's a call from a Siemens SX762 (using ALAW) to FS (no transcoding) which bridges it to a mediagateway. Proxy-media is disabled on the incoming sip_profile. Both au files are extracted from the same pcap file. As you can hear, the second sample sounds bad.. Are there any flags I can set in the incoming or outgoing sip_profile that may fix this ? The server has zero load (it's a Dell R300 with 2.6.24-23-server 64 bit kernel). In fact, this was the only call going through.. FS is version 12163M thanks, Leon On Mar 19, 2009, at 2:15 PM, Brian West wrote: > You shouldn't use it. It has a special use case and I suspect yours > isn't it. Are you doing anything with T.38 right now? > > /b > > On Mar 19, 2009, at 6:57 AM, Leon de Rooij wrote: > >> I'm still undecided yet whether I need proxy-media or not. As I >> understand it, the only downside of enabling proxy-media is that >> early- >> media is not possible, correct ? (Or are there other reasons why I >> shouldn't use proxy-media ?) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Mar 19 08:42:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 10:42:02 -0500 Subject: [Freeswitch-users] Proxy media hickups in audio In-Reply-To: <2E45931E-E495-4DA5-BCA1-8FB545BA2D61@scarlet-internet.nl> References: <2E45931E-E495-4DA5-BCA1-8FB545BA2D61@scarlet-internet.nl> Message-ID: I would have to have the raw pcap to make any sense out of it. /b On Mar 19, 2009, at 10:34 AM, Leon de Rooij wrote: > Brian, > > I put two au files here: > > http://www.ldr.scarlet.nl/ua-to-fs.au > http://www.ldr.scarlet.nl/fs-to-mgw.au > > It's a call from a Siemens SX762 (using ALAW) to FS (no transcoding) > which bridges it to a mediagateway. > Proxy-media is disabled on the incoming sip_profile. > Both au files are extracted from the same pcap file. > > As you can hear, the second sample sounds bad.. Are there any flags I > can set in the incoming or outgoing sip_profile that may fix this ? > > The server has zero load (it's a Dell R300 with 2.6.24-23-server 64 > bit kernel). In fact, this was the only call going through.. > > FS is version 12163M > > thanks, > > Leon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/3c38c007/attachment-0002.html From msc at freeswitch.org Thu Mar 19 08:48:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Mar 2009 08:48:00 -0700 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <8CB7658C3800442-84C-1EAB@WEBMAIL-DY38.sysops.aol.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com> <35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com> <8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com> <35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com> <8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com> <87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com> <87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com> <8CB7658C3800442-84C-1EAB@WEBMAIL-DY38.sysops.aol.com> Message-ID: <87f2f3b90903190848p3839519ah8fa01361c7dfdf84@mail.gmail.com> > tone_detect! sounds good. > > BTW, was there any errors in those extensions I posted. I modified something > you posted MC. Not at first glance. What did you change? -MC From msc at freeswitch.org Thu Mar 19 09:11:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Mar 2009 09:11:36 -0700 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL> <49C12CD8.7020203@gmx.net> <11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <87f2f3b90903190911s6082877at1231c27f6a86506@mail.gmail.com> On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron wrote: > So the second issue is possibly known - really could do with a fix or a > workaround for this as we plan to use E1's for all incoming traffic. > > Can anyone shed light on the first problem (extension rings for a > fraction of a second then hangs up) I mentioned below, or is that > possibly part of the same issue? I have experienced this before but I believe it was resolved by having the telco switch protocol dialects which is probably not an option for you. I think your best bet is to use ozmod_libpri and see if the issue is still present. -MC From msc at freeswitch.org Thu Mar 19 09:14:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Mar 2009 09:14:36 -0700 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: <7.0.1.0.2.20090319125259.02475630@free.fr> References: <7.0.1.0.2.20090319125259.02475630@free.fr> Message-ID: <87f2f3b90903190914o2098b2c2l3b05812b30d38880@mail.gmail.com> On Thu, Mar 19, 2009 at 4:54 AM, Gilles wrote: > Michael Jerris > There is currently no openzap (sangoma, etc) support > on windows, we ?hope this will be coming soon. > > I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper too. > > Would you say the Windows port of Freeswitch is ready to be used > commercially, or I should go for a Linux box instead? Is there a compelling reason to use a Windows machine? If it's a matter of your comfort level with the OS then that's a pretty good reason, especially if you're the one doing support. :) However, I think all of our "power users" are running Linux, and most of them are using 64 bit CentOS on 64 bit hardware. I suppose it all boils down to what you hope to accomplish. The larger the install, the more I would recommend Linux over Windows. -MC From mike at jerris.com Thu Mar 19 09:34:55 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 19 Mar 2009 12:34:55 -0400 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: <7.0.1.0.2.20090319125259.02475630@free.fr> References: <7.0.1.0.2.20090319125259.02475630@free.fr> Message-ID: <40265B71-2148-4296-956B-AA475CC8C267@jerris.com> On Mar 19, 2009, at 7:54 AM, Gilles wrote: > Michael Jerris > There is currently no openzap (sangoma, etc) support > on windows, we hope this will be coming soon. > > I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper > too. > > Would you say the Windows port of Freeswitch is ready to be used > commercially, or I should go for a Linux box instead? > Generally I won't make a recommendation on things like this beyond saying that regardless of what you choose, you should properly test it and verify that it is stable and suitable for your purposes. Windows may work fine for you, it may not, the same goes for linux or anything else, the only way to know for sure is to try it and see if it fits your needs or not. Mike From Mark.Tabron at rnid-typetalk.org.uk Thu Mar 19 09:54:31 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Thu, 19 Mar 2009 16:54:31 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL><49C12CD8.7020203@gmx.net><11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL> <87f2f3b90903190911s6082877at1231c27f6a86506@mail.gmail.com> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765381@tt-mail.RNID.TYPETALK.LOCAL> Thanks, found an install guide on the FS Wiki for libpri - will get the server cloned then install and test. Shall report back. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 19 March 2009 16:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron wrote: > So the second issue is possibly known - really could do with a fix or a > workaround for this as we plan to use E1's for all incoming traffic. > > Can anyone shed light on the first problem (extension rings for a > fraction of a second then hangs up) I mentioned below, or is that > possibly part of the same issue? I have experienced this before but I believe it was resolved by having the telco switch protocol dialects which is probably not an option for you. I think your best bet is to use ozmod_libpri and see if the issue is still present. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- From mszlazak at aol.com Thu Mar 19 10:14:38 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 19 Mar 2009 13:14:38 -0400 Subject: [Freeswitch-users] Is mod vmd working? In-Reply-To: <87f2f3b90903190848p3839519ah8fa01361c7dfdf84@mail.gmail.com> References: <8CB75AE46C9F7D8-56C-D54@webmail-mf21.sysops.aol.com><35b355e90903180011x45ff5671me076623149b41245@mail.gmail.com><8CB75D8D844C301-358-E82@webmail-dh10.sysops.aol.com><35b355e90903180607k377221d2w1f7bb2f50271364a@mail.gmail.com><8CB75F5E35EE3F6-C80-1841@webmail-dx06.sysops.aol.com><87f2f3b90903181024h2d3964b6xd4bd9299ff719a66@mail.gmail.com><87f2f3b90903181715h2e50bab4n61f25fe57833147e@mail.gmail.com><8CB7658C3800442-84C-1EAB@WEBMAIL-DY38.sysops.aol.com> <87f2f3b90903190848p3839519ah8fa01361c7dfdf84@mail.gmail.com> Message-ID: <8CB76CD515B680C-418-AB@WEBMAIL-DZ07.sysops.aol.com> I put this after the "vmd" tag to check vmd with tones found on this page http://en.wikipedia.org/wiki/Special_information_tone I converted them over with Audacity to wav files and "vmd" worked in finding a "beep" but the format was wrong for FS. However, after I switch the format of the audio files to something FS likes then vmd would not detect the tones. Is there some good test tones for the U.S phone system I could use to check both mod_vmd and tone_detect? Thanks. Mark. -----Original Message-----From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Thu, 19 Mar 2009 8:48 am Subject: Re: [Freeswitch-users] Is mod vmd working? > tone_detect! sounds good. > > BTW, was there any errors in those extensions I posted. I modified something > you posted MC. Not at first glance. What did you change? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/0bf36af8/attachment-0002.html From lukasz at czerpak.eu Thu Mar 19 11:22:07 2009 From: lukasz at czerpak.eu (=?UTF-8?B?xYF1a2FzeiBDemVycGFr?=) Date: Thu, 19 Mar 2009 19:22:07 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) Message-ID: <49C28D4F.4040307@czerpak.eu> Hi, I have some troubles with provider configuration. The are warnings in logs: 2009-03-19 19:02:48 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 40 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. My provider uses SER and Cisco hardware (Cisco-SIPGateway/IOS-12.x). Moreover, above warning appears only when someone calls to FreeSWITCH (voice quality is poor) - connection from FreeSWITCH works without any warnings and has perfect quality. Other my providers works perfect. Is there any solution of this problem? regards, Lukasz From brian at freeswitch.org Thu Mar 19 11:28:44 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 13:28:44 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C28D4F.4040307@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> Message-ID: <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> Try: /b On Mar 19, 2009, at 1:22 PM, ?ukasz Czerpak wrote: > Is there any solution of this problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/88211d93/attachment-0002.html From lukasz at czerpak.eu Thu Mar 19 11:52:22 2009 From: lukasz at czerpak.eu (=?UTF-8?B?xYF1a2FzeiBDemVycGFr?=) Date: Thu, 19 Mar 2009 19:52:22 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> References: <49C28D4F.4040307@czerpak.eu> <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> Message-ID: <49C29466.5040708@czerpak.eu> Brian West wrote: > Try: > > > * Unfortunately there is no difference when it is set to 'scrooge' or other value :( regards, Lukasz From brian at freeswitch.org Thu Mar 19 11:58:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 13:58:06 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C29466.5040708@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> <49C29466.5040708@czerpak.eu> Message-ID: <3B28A868-3ACE-4BFB-80D0-6D89E7384BDF@freeswitch.org> what rev are you on? /b On Mar 19, 2009, at 1:52 PM, ?ukasz Czerpak wrote: >> * > > Unfortunately there is no difference when it is set to 'scrooge' or > other value :( > > > regards, > Lukasz From kristian.kielhofner at gmail.com Thu Mar 19 11:59:03 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 19 Mar 2009 14:59:03 -0400 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C28D4F.4040307@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> Message-ID: <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> Hi, This is a known issue with some of these platforms but for completeness can you send the actual SDP? 2009/3/19 ?ukasz Czerpak : > Hi, > > I have some troubles with provider configuration. The are warnings in logs: > > 2009-03-19 19:02:48 [WARNING] mod_sofia.c:739 sofia_read_frame() We were > told to use ptime 20 but what they meant to say was 40 > > This issue has so far been identified to happen on the following broken > platforms/devices: > > Linksys/Sipura aka Cisco > > > ShoreTel > > > Sonus/L3 > > > We will try to fix it but some of the devices on this list are so broken > who knows what will happen.. > > My provider uses SER and Cisco hardware (Cisco-SIPGateway/IOS-12.x). > > Moreover, above warning appears only when someone calls to FreeSWITCH > (voice quality is poor) - connection from FreeSWITCH works without any > warnings and has perfect quality. > > Other my providers works perfect. > > Is there any solution of this problem? > > regards, > Lukasz > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From lukasz at czerpak.eu Thu Mar 19 12:20:18 2009 From: lukasz at czerpak.eu (=?ISO-8859-2?Q?=A3ukasz_Czerpak?=) Date: Thu, 19 Mar 2009 20:20:18 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> Message-ID: <49C29AF2.2000703@czerpak.eu> Kristian Kielhofner wrote: > Hi, > > This is a known issue with some of these platforms but for > completeness can you send the actual SDP? My voip configuration: +------------+ +----------+ | ROUTER | LAN | Linksys | NET----| with |---------| PAP2T-EU | | FreeSWITCH | | | +------------+ +----------+ Linksys has changed rtp packet size to 0.020 On FreeSWITCH i am using g729 (http://freehg.org/u/deepwalker/fs_g729/) - for testing only. SDP logs are below, full session's log is in attachment: 2009-03-19 20:07:02 [DEBUG] sofia.c:2806 sofia_handle_sip_i_state() Channel sofia/external/0607xxxxxx at 217.11.128.50 entering state [received] 2009-03-19 20:07:02 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=CiscoSystemsSIP-GW-UserAgent 9221 3981 IN IP4 217.11.128.50 s=SIP Call c=IN IP4 217.11.128.50 t=0 0 m=audio 18292 RTP/AVP 18 4 2 98 99 0 8 3 100 101 19 c=IN IP4 217.11.128.50 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=rtpmap:2 G726-32/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:99 G726-16/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=direction:active ... 2009-03-19 20:07:02 [INFO] mod_sofia.c:1351 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1237472006 1237472007 IN IP4 89.79.191.29 s=FreeSWITCH c=IN IP4 89.79.191.29 t=0 0 m=audio 17616 RTP/AVP 18 101 19 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20 a=sendrecv ... 2009-03-19 20:07:03 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=- 121150237 121150237 IN IP4 192.168.1.250 s=- c=IN IP4 192.168.1.250 t=0 0 m=audio 16438 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 ... 2009-03-19 20:07:03 [DEBUG] mod_sofia.c:503 sofia_answer_channel() Local SDP sofia/external/0607320038 at 217.11.128.50: v=0 o=FreeSWITCH 1237472006 1237472008 IN IP4 89.79.191.29 s=FreeSWITCH c=IN IP4 89.79.191.29 t=0 0 m=audio 17616 RTP/AVP 18 101 19 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20 a=sendrecv ... 2009-03-19 20:07:04 [DEBUG] sofia.c:2810 sofia_handle_sip_i_state() Remote SDP: v=0 o=CiscoSystemsSIP-GW-UserAgent 9221 3981 IN IP4 217.11.128.50 s=SIP Call c=IN IP4 217.11.128.50 t=0 0 m=audio 18292 RTP/AVP 18 19 101 100 c=IN IP4 217.11.128.50 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=ptime:60 regards, Lukasz -------------- next part -------------- A non-text attachment was scrubbed... Name: session2.log Type: text/x-log Size: 40912 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/816d2158/attachment-0002.bin From brian at freeswitch.org Thu Mar 19 12:27:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 14:27:56 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C29AF2.2000703@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> Message-ID: Well you can't have ptime 60 one way and 20 the other it just won't work. Also I can't even think that this illegal codec was even tested at 60ms... Try it with ulaw and see what it does. or only allow G729 at 60i and see what it does. /b On Mar 19, 2009, at 2:20 PM, ?ukasz Czerpak wrote: > Linksys has changed rtp packet size to 0.020 > On FreeSWITCH i am using g729 (http://freehg.org/u/deepwalker/ > fs_g729/) - for testing only. > > SDP logs are below, full session's log is in attachment: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/79d48f4d/attachment-0002.html From steve.d.ward at gmail.com Thu Mar 19 12:42:55 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Thu, 19 Mar 2009 15:42:55 -0400 Subject: [Freeswitch-users] not hanging up Message-ID: <4ea6e8f20903191242vb56bcaetb7acd4853b215b0e@mail.gmail.com> I have phones registered to a FS box, and an * box. There is a sip trunk between the two boxes. A phone on my * (54321) calls a FS phone (12345); if I hang up the * phone while it's still ringing, this is what I get on the sip trace on FS: ... 2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/12345 at 11.2.22.45! recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950: ------------------------------------------------------------------------ CANCEL sip:12345 at b-pbx-sip-3.abc.xyz.netSIP/2.0 Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport From: "Steve" >;tag=as25193d44 To: > Call-ID: 0c0614d866a62841546cbf3340224682 at 11.2.22.45 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 328 bytes to udp/[11.2.22.45]:5060 at 19:05:44.313572: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport=5060 From: "Steve" >;tag=as25193d44 To: >;tag=c5Z8Q1e93p7KD Call-ID: 0c0614d866a62841546cbf3340224682 at 11.2.22.45 CSeq: 103 CANCEL Content-Length: 0 -------------------------------------------------------- The effect is that the FS keeps on ringing - it doesn't detect the hangup. When I call from a FS phone (1000) to another FS phone (12345), and I hang up the calling phone while it's still ringing, this is what I get on the sip trace: ... send 425 bytes to udp/[11.2.56.106]:63054 at 19:15:29.737163: ------------------------------------------------------------------------ CANCEL sip:12345 at 11.2.56.106:63054;rinstance=64e968d7a1317bc3 SIP/2.0 Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a Max-Forwards: 69 From: "Extension 1000" >;tag=meK8yUgpgU2Zc To: Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" Content-Length: 0 ------------------------------------------------------------------------ recv 427 bytes from udp/[11.2.56.106]:63054 at 19:15:29.838863: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a Contact: To: ;tag=db12c87a From: "Extension 1000" >;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 CANCEL User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ------------------------------------------------------------------------ recv 376 bytes from udp/[11.2.56.106]:63054 at 19:15:29.839334: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a To: ;tag=db12c87a From: "Extension 1000" >;tag=meK8yUgpgU2Zc Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 CSeq: 112626727 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ... It works just fine. Any ideas? I'm not sure where to go with this. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/67594f9f/attachment-0002.html From lukasz at czerpak.eu Thu Mar 19 12:43:51 2009 From: lukasz at czerpak.eu (=?UTF-8?B?xYF1a2FzeiBDemVycGFr?=) Date: Thu, 19 Mar 2009 20:43:51 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <3B28A868-3ACE-4BFB-80D0-6D89E7384BDF@freeswitch.org> References: <49C28D4F.4040307@czerpak.eu> <85ACCAE8-D041-48BB-B50C-75946D8A0CD4@freeswitch.org> <49C29466.5040708@czerpak.eu> <3B28A868-3ACE-4BFB-80D0-6D89E7384BDF@freeswitch.org> Message-ID: <49C2A077.8080409@czerpak.eu> Brian West wrote: > what rev are you on? > trunk - ~2009-03-15 21:00 regards, ?ukasz From lukasz at czerpak.eu Thu Mar 19 13:06:31 2009 From: lukasz at czerpak.eu (=?ISO-8859-2?Q?=A3ukasz_Czerpak?=) Date: Thu, 19 Mar 2009 21:06:31 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> Message-ID: <49C2A5C7.10205@czerpak.eu> Brian West wrote: > Well you can't have ptime 60 one way and 20 the other it just won't > work. Also I can't even think that this illegal codec was even tested > at 60ms... Try it with ulaw and see what it does. or only allow > G729 at 60i and see what it does. > I've just tested G729 at 60i and everything works perfect - thank you very much. I didn't test ulaw. What is wrong - my provider is incompatible with specification or FreeSWITCH has problem with codec negotiation? Is there any possibility to force codec for specific gateway/provider? regards Lukasz From brian at freeswitch.org Thu Mar 19 13:31:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 15:31:18 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C2A5C7.10205@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> <49C2A5C7.10205@czerpak.eu> Message-ID: The issue I seen was they invite to you with NO ptime which indicates 20ms, they should invite with ptime:60 if they want 60. /b On Mar 19, 2009, at 3:06 PM, ?ukasz Czerpak wrote: > > I've just tested G729 at 60i and everything works perfect - thank you > very > much. I didn't test ulaw. > What is wrong - my provider is incompatible with specification or > FreeSWITCH has problem with codec negotiation? > > Is there any possibility to force codec for specific gateway/provider? > > regards > Lukasz From qulix at mail.ru Thu Mar 19 13:48:32 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Thu, 19 Mar 2009 23:48:32 +0300 Subject: [Freeswitch-users] =?koi8-r?b?R29vZCB0aW1lLCBwZW9wbGUh?= Message-ID: Thats the thing!! Im using tcpdump to watch for packets - and i dont see any mistakes =\ The xml i sent is allright, its like a piece from my static worked xml dialplan. But I cant understand why does FS recognise it as a 130+ mb file :D Maybe i need to update s0mthing?) Brian West ?????: > Any reason you're feeding it a 130+ meg file over XML_CURL? > > /b > > On Mar 19, 2009, at 6:05 AM, ????... wrote: > >> Hello! >> >> Has anybody faced such a problem with xml_curl? >> 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing 1000->********** in context default >> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() >> Oversized file detected [136089828 bytes] >> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error >> encountered! >> >> Tcpdump log tells that transaction is allright (xml dialplan is fine >> and etc) >> But FS says its oversized =\ what could be wrong? >> >> My trunk is : >> FreeSWITCH Version 1.0.trunk (12573). > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Mar 19 13:52:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 15:52:22 -0500 Subject: [Freeswitch-users] Good time, people! In-Reply-To: References: Message-ID: <9A3D7EE5-FA78-4C64-9868-6E46AB12ED5A@freeswitch.org> can you do a raw request with wget? /b On Mar 19, 2009, at 3:48 PM, ????... wrote: > Thats the thing!! > Im using tcpdump to watch for packets - and i dont see any mistakes =\ > The xml i sent is allright, its like a piece from my static worked > xml dialplan. > > But I cant understand why does FS recognise it as a 130+ mb file :D > Maybe i need to update s0mthing?) From lukasz at czerpak.eu Thu Mar 19 14:03:13 2009 From: lukasz at czerpak.eu (=?UTF-8?B?xYF1a2FzeiBDemVycGFr?=) Date: Thu, 19 Mar 2009 22:03:13 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> <49C2A5C7.10205@czerpak.eu> Message-ID: <49C2B311.2040105@czerpak.eu> Brian West wrote: > The issue I seen was they invite to you with NO ptime which indicates > 20ms, they should invite with ptime:60 if they want 60. > I see but there is any solution to bypass this provider's "incompatibility"? I want to stay with this provider anyway - he has very good quality and nice prices ;) thanks and regards, ?ukasz From brian at freeswitch.org Thu Mar 19 14:08:46 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Mar 2009 16:08:46 -0500 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C2B311.2040105@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> <49C2A5C7.10205@czerpak.eu> <49C2B311.2040105@czerpak.eu> Message-ID: Well you said you were using G.729 for testing... when you're clearly not... but I told you already how to fix it... for that IP or peer G729 at 60i /b On Mar 19, 2009, at 4:03 PM, ?ukasz Czerpak wrote: > I see but there is any solution to bypass this provider's > "incompatibility"? I want to stay with this provider anyway - he has > very good quality and nice prices ;) > > thanks and regards, > ?ukasz From qulix at mail.ru Thu Mar 19 14:28:45 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Fri, 20 Mar 2009 00:28:45 +0300 Subject: [Freeswitch-users] =?koi8-r?b?R29vZCB0aW1lLCBwZW9wbGUh?= Message-ID: Its not the easy thing. But what I can do is to attach here full tcpdump log, with all packets. Brian West ?????: > can you do a raw request with wget? > > /b > > On Mar 19, 2009, at 3:48 PM, ????... wrote: > >> Thats the thing!! >> Im using tcpdump to watch for packets - and i dont see any mistakes =\ >> The xml i sent is allright, its like a piece from my static worked >> xml dialplan. >> >> But I cant understand why does FS recognise it as a 130+ mb file :D >> Maybe i need to update s0mthing?) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sprice at gmail.com Thu Mar 19 14:33:58 2009 From: sprice at gmail.com (SP) Date: Thu, 19 Mar 2009 16:33:58 -0500 Subject: [Freeswitch-users] Good time, people! In-Reply-To: References: Message-ID: <7e2ac3270903191433h35e043d3l8e948690599a609@mail.gmail.com> Are you setting a Content Length header in the HTTP response?? 2009/3/19 ????... : > Thats the thing!! > Im using tcpdump to watch for packets - and i dont see any mistakes =\ > The xml i sent is allright, its like a piece from my static worked xml dialplan. > > But I cant understand why does FS recognise it as a 130+ mb file :D > Maybe i need to update s0mthing?) > > Brian West ?????: >> Any reason you're feeding it a 130+ meg file over XML_CURL? >> >> /b >> >> On Mar 19, 2009, at 6:05 AM, ????... wrote: >> >>> Hello! >>> >>> Has anybody faced such a problem with xml_curl? >>> 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>> Processing 1000->********** in context default >>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() >>> Oversized file detected [136089828 bytes] >>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error >>> encountered! >>> >>> Tcpdump log tells that transaction is allright (xml dialplan is fine >>> and etc) >>> But FS says its oversized =\ what could be wrong? >>> >>> My trunk is : >>> FreeSWITCH Version 1.0.trunk (12573). >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From qulix at mail.ru Thu Mar 19 14:45:17 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Fri, 20 Mar 2009 00:45:17 +0300 Subject: [Freeswitch-users] =?koi8-r?b?R29vZCB0aW1lLCBwZW9wbGUh?= Message-ID: As I see theres only : Content-Type: text/html; charset=utf-8 But no Content Length SP ?????: > Are you setting a Content Length header in the HTTP response?? > > 2009/3/19 ????... : >> Thats the thing!! >> Im using tcpdump to watch for packets - and i dont see any mistakes =\ >> The xml i sent is allright, its like a piece from my static worked xml dialplan. >> >> But I cant understand why does FS recognise it as a 130+ mb file :D >> Maybe i need to update s0mthing?) >> >> Brian West ?????: >>> Any reason you're feeding it a 130+ meg file over XML_CURL? >>> >>> /b >>> >>> On Mar 19, 2009, at 6:05 AM, ????... wrote: >>> >>>> Hello! >>>> >>>> Has anybody faced such a problem with xml_curl? >>>> 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>>> Processing 1000->********** in context default >>>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback() >>>> Oversized file detected [136089828 bytes] >>>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error >>>> encountered! >>>> >>>> Tcpdump log tells that transaction is allright (xml dialplan is fine >>>> and etc) >>>> But FS says its oversized =\ what could be wrong? >>>> >>>> My trunk is : >>>> FreeSWITCH Version 1.0.trunk (12573). >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From codecomplete at free.fr Thu Mar 19 15:45:07 2009 From: codecomplete at free.fr (Gilles) Date: Thu, 19 Mar 2009 23:45:07 +0100 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? Message-ID: <7.0.1.0.2.20090319232207.0242a5d8@fredshack.com> (sorry for the broken thread: I don't know how to avoid this when answering through the digest version of the mailing list) Michael Jerris > You could use Netborder Express with it. Thanks for the tip. I didn't know this device. I'm not sure I understand the difference between this PCI card and other Sangoma PCI cards that offer an FXO port, though :-/ mercutioviz > Is there a compelling reason to use a Windows machine? Yes. I'd like to offer a really cheap solution for those customers who don't mind using their workstation as Freeswitch IVR server, so I can just provide a Linksys VoIP gateway and the software for Windows, and they're ready to go. I'll go ahead and play with the Windows port of Freeswitch, and see how it goes. Thank you. From mszlazak at aol.com Thu Mar 19 16:35:47 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Thu, 19 Mar 2009 19:35:47 -0400 Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? In-Reply-To: <7.0.1.0.2.20090319232207.0242a5d8@fredshack.com> References: <7.0.1.0.2.20090319232207.0242a5d8@fredshack.com> Message-ID: <8CB7702907EC8A8-1188-10E9@WEBMAIL-DY32.sysops.aol.com> I'm doing what you want to do and using SPA3102.? It's much easier to get someone to try it this way when dealing with small mom and pop size business.? Haven't tried higher concurrent call volumes with some of the PCI cards mentioned. If you haven't done this already, my advice is first to see whether there is enough of a market for you and what the issues will be with potential customers before investing to much time on the technical side. That means putting on your suit on and visiting businesses for a few months with a notebook. I'll tell you some things, first their maybe many businesses that don't want an internet connection and don't even bother mentioning voip if you want to resell that service. The rest you'll find out. Good luck. Mark. -----Original Message----- From: Gilles To: freeswitch-users at lists.freeswitch.org Sent: Thu, 19 Mar 2009 3:45 pm Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? (sorry for the broken thread: I don't know how to avoid this when answering through the digest version of the mailing list) Michael Jerris > You could use Netborder Express with it. Thanks for the tip. I didn't know this device. I'm not sure I understand the difference between this PCI card and other Sangoma PCI cards that offer an FXO port, though :-/ mercutioviz > Is there a compelling reason to use a Windows machine? Yes. I'd like to offer a really cheap solution for those customers who don' t mind using their workstation as Freeswitch IVR server, so I can just provide a Linksys VoIP gateway and the software for Windows, and they're ready to go. I'll go ahead and play with the Windows port of Freeswitch, and see how it goes. Thank you. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090319/13415cb2/attachment-0002.html From switchserver at gmail.com Thu Mar 19 19:53:29 2009 From: switchserver at gmail.com (HarryK) Date: Thu, 19 Mar 2009 19:53:29 -0700 (PDT) Subject: [Freeswitch-users] Skypiax and no-audio (NAT) issue Message-ID: <22613477.post@talk.nabble.com> Ok I got Skypiax working just fine but there is no audio either way when I say call into a conf using the Skype username. I had this no audio problem with NAT when I first setup FreeSWITCH and solved it by using "Scenario 2" from this wiki page... http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios Now I'm stumped because I THINK I have to traverse NAT once again but not seeing anyway to tell Skypiax that it must use the "doublenat" profile! If thats even close to the solution I'm not sure. The call is ending up in the proper conf, just no audio. I've confirmed this. Does someone have any experience with what I'm dealing with? Thanks -- View this message in context: http://www.nabble.com/Skypiax-and-no-audio-%28NAT%29-issue-tp22613477p22613477.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From switchserver at gmail.com Thu Mar 19 19:55:58 2009 From: switchserver at gmail.com (HarryK) Date: Thu, 19 Mar 2009 19:55:58 -0700 (PDT) Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <49C239D7.4080101@freeswitch.org> References: <22594923.post@talk.nabble.com> <49C239D7.4080101@freeswitch.org> Message-ID: <22613496.post@talk.nabble.com> I guess that's why they call us noobs! heh ;) Working perfectly, thank you!! Raymond Chandler-2 wrote: > > HarryK wrote: >> I have Cepstral working. >> >> Can someone please tell me how to go about having it read RSS feeds? I >> can >> have the dialplan direct it np. But I really dont have a clue how to >> point >> it at an RSS. Any help would be great, ddint find anything in the wiki. >> >> >> > have you tried mod_rss? > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22613496.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Thu Mar 19 20:26:53 2009 From: dujinfang at gmail.com (seven) Date: Fri, 20 Mar 2009 11:26:53 +0800 Subject: [Freeswitch-users] Skypiax and no-audio (NAT) issue In-Reply-To: <22613477.post@talk.nabble.com> References: <22613477.post@talk.nabble.com> Message-ID: <86005AC0-14AE-4505-B97D-2A6F3F2B0A22@gmail.com> I don't think no sound is caused by NAT, better to check sound driver and configuration. On Mar 20, 2009, at 10:53 AM, HarryK wrote: > > Ok I got Skypiax working just fine but there is no audio either way > when I > say call into a conf using the Skype username. > > I had this no audio problem with NAT when I first setup FreeSWITCH and > solved it by using "Scenario 2" from this wiki page... > > http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios > > Now I'm stumped because I THINK I have to traverse NAT once again > but not > seeing anyway to tell Skypiax that it must use the "doublenat" > profile! If > thats even close to the solution I'm not sure. The call is ending up > in the > proper conf, just no audio. I've confirmed this. > > Does someone have any experience with what I'm dealing with? > > > Thanks > -- > View this message in context: http://www.nabble.com/Skypiax-and-no-audio-%28NAT%29-issue-tp22613477p22613477.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mattdfong at gmail.com Thu Mar 19 20:46:45 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 20 Mar 2009 10:46:45 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> Message-ID: <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> Hi Anthony, I installed the patch, but I don't think it accomplishes what I want. I want the opposite, I want the fifo caller to continue along with the dialplan after the agent (consumer) is finished with servicing the call. This might be useful in situations where there could be an IVR recording customer satisfaction results of the agent servicing the call. As is, FS ends the caller's channel after finishing up in the fifo (ie, agent (consumer) disconnects or hangsup)--there should be life after s/he has been serviced by an agent (preferably continuing on in the dialplan). If I'm confused and missing something obvious, please correct me... Thanks --matt 2009/3/19 Anthony Minessale > This is the patch > > http://jira.freeswitch.org/browse/MODAPP-237 > > it's not added yet. > > > 2009/3/18 Matthew Fong > > I upgraded to >> FreeSWITCH Version 1.0.trunk (12654M) >> >> but caller is still being hungup (and not continuing on with dialplan) >> after agent disconnect with hangup_after_bridge=false >> >> Is there a separate patch I need to apply? Thanks. >> >> --matt >> >> >> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >> >>> Hi Anthony, thanks for the reply. >>> I've searched thru jira, and didn't find anything when searching for fifo >>> that was recently updated or related, except >>> >>> http://jira.freeswitch.org/browse/MODAPP-189 >>> >>> and I'm not sure if this does what I need. Was this what you were >>> referring to? Thanks. >>> >>> --matt >>> >>> 2009/3/17 Anthony Minessale >>> >>> there is a patch in jira that will implement this feature about to be >>>> added >>>> >>>> >>>> 2009/3/17 Matthew Fong >>>> >>>>> I apologize if this is a double post to -dev. I'm not sure why I don't >>>>> see my message appearing, so I'm going to try again in the -user list (first >>>>> timer posting here ;). >>>>> >>>>> I have a situation where it would be useful for a caller not to be >>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>> situation. It would be preferable if the caller channel went further along >>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>> minimal. I also tried executing fifo in a lua app and >>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>> could be done as a feature enhancement? Thanks. >>>>> >>>>> --matt >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/5e72790d/attachment-0002.html From carlos.talbot at gmail.com Thu Mar 19 22:02:04 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Fri, 20 Mar 2009 00:02:04 -0500 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <22594923.post@talk.nabble.com> References: <22594923.post@talk.nabble.com> Message-ID: <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> I wrote this wiki page a while back. Did it help? http://wiki.freeswitch.org/wiki/Mod_rss On Thu, Mar 19, 2009 at 2:41 AM, HarryK wrote: > > I have Cepstral working. > > Can someone please tell me how to go about having it read RSS feeds? I can > have the dialplan direct it np. But I really dont have a clue how to point > it at an RSS. Any help would be great, ddint find anything in the wiki. > > > -- > View this message in context: > http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22594923.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/2fd8228d/attachment-0002.html From brian at freeswitch.org Thu Mar 19 22:08:23 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Mar 2009 00:08:23 -0500 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> Message-ID: http://jira.freeswitch.org/browse/MODASRTTS-11 Might wanna know about that issue also :) /b On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote: > I wrote this wiki page a while back. Did it help? > > http://wiki.freeswitch.org/wiki/Mod_rss -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/935dabec/attachment-0002.html From lukasz at czerpak.eu Fri Mar 20 01:05:40 2009 From: lukasz at czerpak.eu (=?ISO-8859-2?Q?=A3ukasz_Czerpak?=) Date: Fri, 20 Mar 2009 09:05:40 +0100 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> Message-ID: <49C34E54.9000000@czerpak.eu> Brian West pisze: > Well you can't have ptime 60 one way and 20 the other it just won't > work. Also I can't even think that this illegal codec was even tested > at 60ms... Try it with ulaw and see what it does. or only allow > G729 at 60i and see what it does. > I made some tests with ulaw with success. There is no problem with ptime negotiation. Will the g729 codec be fully (not passthrough) supported in FreeSWITCH? regards, -- ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] From andy at fabulous4.co.uk Fri Mar 20 01:40:41 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Fri, 20 Mar 2009 08:40:41 -0000 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> Message-ID: <153785972B6D42C99E8CAF7DE2F1A145@wsandy> Thanks Brian, I've upgraded to 1.0.3 and things seem a little better but I'm still loosing the gateway connection intermittently. I rebuilt the config on upgrade, is there any possibility I've missed something? Is there a keep-alive setting for a gateway or a re-connect after x or something. Many thanks for your help. Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 18 March 2009 14:08 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Losing Gateway registration Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/bd4029fd/attachment-0002.html From ludovic.fouquet at bewan.com Fri Mar 20 03:11:08 2009 From: ludovic.fouquet at bewan.com (ludovic) Date: Fri, 20 Mar 2009 11:11:08 +0100 Subject: [Freeswitch-users] SIP registration fails when using hostname in sip_profile ? In-Reply-To: <246296A5-851A-4859-BCA9-05E2415A20EA@freeswitch.org> References: <49BE928D.3090509@bewan.com> <49BF87E5.5090809@bewan.com> <49BFE901.3070709@bewan.com> <246296A5-851A-4859-BCA9-05E2415A20EA@freeswitch.org> Message-ID: <49C36BBC.1060708@bewan.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/f8f5ac6a/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: bewan100.jpg Type: image/jpeg Size: 3963 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/f8f5ac6a/attachment-0002.jpg From anthony.minessale at gmail.com Fri Mar 20 05:04:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Mar 2009 07:04:09 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> Message-ID: <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> The agent could transfer the caller to another extension. 2009/3/19 Matthew Fong > Hi Anthony, > I installed the patch, but I don't think it accomplishes what I want. > > I want the opposite, I want the fifo caller to continue along with the > dialplan after the agent (consumer) is finished with servicing the call. > This might be useful in situations where there could be an IVR recording > customer satisfaction results of the agent servicing the call. As is, FS > ends the caller's channel after finishing up in the fifo (ie, agent > (consumer) disconnects or hangsup)--there should be life after s/he has been > serviced by an agent (preferably continuing on in the dialplan). > > If I'm confused and missing something obvious, please correct me... Thanks > > --matt > > > > 2009/3/19 Anthony Minessale > > This is the patch >> >> http://jira.freeswitch.org/browse/MODAPP-237 >> >> it's not added yet. >> >> >> 2009/3/18 Matthew Fong >> >> I upgraded to >>> FreeSWITCH Version 1.0.trunk (12654M) >>> >>> but caller is still being hungup (and not continuing on with dialplan) >>> after agent disconnect with hangup_after_bridge=false >>> >>> Is there a separate patch I need to apply? Thanks. >>> >>> --matt >>> >>> >>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>> >>>> Hi Anthony, thanks for the reply. >>>> I've searched thru jira, and didn't find anything when searching for >>>> fifo that was recently updated or related, except >>>> >>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>> >>>> and I'm not sure if this does what I need. Was this what you were >>>> referring to? Thanks. >>>> >>>> --matt >>>> >>>> 2009/3/17 Anthony Minessale >>>> >>>> there is a patch in jira that will implement this feature about to be >>>>> added >>>>> >>>>> >>>>> 2009/3/17 Matthew Fong >>>>> >>>>>> I apologize if this is a double post to -dev. I'm not sure why I don't >>>>>> see my message appearing, so I'm going to try again in the -user list (first >>>>>> timer posting here ;). >>>>>> >>>>>> I have a situation where it would be useful for a caller not to be >>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>> situation. It would be preferable if the caller channel went further along >>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>> minimal. I also tried executing fifo in a lua app and >>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>> could be done as a feature enhancement? Thanks. >>>>>> >>>>>> --matt >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/51f08e52/attachment-0002.html From mark at markehle.net Fri Mar 20 05:29:18 2009 From: mark at markehle.net (Mark) Date: Fri, 20 Mar 2009 08:29:18 -0400 Subject: [Freeswitch-users] (OT) SPA-922 unlock Message-ID: <20090320082918.7060959me3mtzbxc@gateway.ehle.homelinux.org> I was never able to unlock these phones using some sort of sniffing/back door stuff. I was, however, lucky enough to have the previous owner discover the password. If it helps anybody, the now-defunct Webnet Global Communications (where the phones came from at a bankruptcy auction) used the username Admin and password 'webnet'. Thanks to all who provided information on this endeavor. Library Mark Quoting Gabriel Kuri : > On the slight chance they're not doing remote provisioning and the phone > is just simply locked with a username/password, you'll need to feed the > phone a TFTP server via DHCP Option 66 and setup a config file on that > tftp server with the name spa922.cfg. > > Contact me off list about generating a config file for the phone. > > Gabe > > Mark wrote: >> I did unplug the ethernet cable. I have never been able to make the >> IVR work on any of the Linksys phones that I have. I must be doing >> something wrong. >> >> I will try to sniff the traffic on the phone when I start it up. I >> will report back when I do. >> >> Thanks so much - >> >> Library Mark >> >> Quoting "Gabriel Kuri" : >> >>> I believe you need to make sure the Ethernet cable is unplugged from the >>> phone when trying to dial that string. >>> >>> Now I've never tried this, but it should theoretically be possible ... >>> >>> Sniff the traffic of the phone and see where it's attempting to pickup >>> the config file. Then setup a local network with your own DNS server, >>> and re-direct the phone (via DNS) to your own web server (assuming it's >>> picking up the config via http) and have a config file on the web server >>> with a username and password you specify to reset the config and get >>> into the phone. Let's hope they didn't setup the phone to provision via >>> https, otherwise you're really SOL >>> >>> If you need help generating a config for the phone, with Linksys' >>> special config tool, contact me offlist. >>> >>> Gabe >>> >>> Mark wrote: >>>> Sadly, ****73738# does not work. >>>> >>>> Is there a jumper on the board or some other hardware fix for this? >>>> >>>> Quoting "Gabriel Kuri" : >>>> >>>>> Have you tried resetting the phone via the built-in IVR menu? >>>>> >>>>> Pick up the handset and dial ****73738# >>>>> >>>>> This should reset the phone to factory defaults, assuming that company >>>>> didn't lock this feature out. >>>>> >>>>> Gabe >>>>> >>>>> >>>>> >>>>> Mark wrote: >>>>>> Hello, folks - I hope that I can reach someone who knows the answer to >>>>>> this one: >>>>>> >>>>>> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >>>>>> locked by Webnet global Communications. From what I can tell, this >>>>>> company went bankrupt, and the ebay seller bought the phones from a >>>>>> bankruptcy auction. He does not know the admin username or password. >>>>>> Nowhere on the linksys site is there a solution to how to unlock these >>>>>> phones. >>>>>> >>>>>> Is there a way, or did I buy 2 interesting looking doorstops? Other >>>>>> than the password thing, they function fine. >>>>>> >>>>>> Thanks - >>>>>> >>>>>> Library Mark >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mattdfong at gmail.com Fri Mar 20 06:04:10 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 20 Mar 2009 20:04:10 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> Message-ID: <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> Hi Anthony, I'm trying to use fifo in a different sense. Instead of using it for inbound call queing, I'd like to use it for outbound call making. So instead, my agents are "waiting" in the que, and once an outbound call is connected, the "caller" will take an agent out of the que. So, in my case, the Fifo agent, would not be able to transfer the call because it's an outbound call, and the phone number on the other side is that of a non-employee. Fifo works a little smoother this way, because in reality, for outbound call making to an agent, this is what's happening, not vica versa. How difficult would this be to implement? Thanks. --matt 2009/3/20 Anthony Minessale > The agent could transfer the caller to another extension. > > > 2009/3/19 Matthew Fong > > Hi Anthony, >> I installed the patch, but I don't think it accomplishes what I want. >> >> I want the opposite, I want the fifo caller to continue along with the >> dialplan after the agent (consumer) is finished with servicing the call. >> This might be useful in situations where there could be an IVR recording >> customer satisfaction results of the agent servicing the call. As is, FS >> ends the caller's channel after finishing up in the fifo (ie, agent >> (consumer) disconnects or hangsup)--there should be life after s/he has been >> serviced by an agent (preferably continuing on in the dialplan). >> >> If I'm confused and missing something obvious, please correct me... Thanks >> >> --matt >> >> >> >> 2009/3/19 Anthony Minessale >> >> This is the patch >>> >>> http://jira.freeswitch.org/browse/MODAPP-237 >>> >>> it's not added yet. >>> >>> >>> 2009/3/18 Matthew Fong >>> >>> I upgraded to >>>> FreeSWITCH Version 1.0.trunk (12654M) >>>> >>>> but caller is still being hungup (and not continuing on with dialplan) >>>> after agent disconnect with hangup_after_bridge=false >>>> >>>> Is there a separate patch I need to apply? Thanks. >>>> >>>> --matt >>>> >>>> >>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>>> >>>>> Hi Anthony, thanks for the reply. >>>>> I've searched thru jira, and didn't find anything when searching for >>>>> fifo that was recently updated or related, except >>>>> >>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>> >>>>> and I'm not sure if this does what I need. Was this what you were >>>>> referring to? Thanks. >>>>> >>>>> --matt >>>>> >>>>> 2009/3/17 Anthony Minessale >>>>> >>>>> there is a patch in jira that will implement this feature about to be >>>>>> added >>>>>> >>>>>> >>>>>> 2009/3/17 Matthew Fong >>>>>> >>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>> (first timer posting here ;). >>>>>>> >>>>>>> I have a situation where it would be useful for a caller not to be >>>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>>> situation. It would be preferable if the caller channel went further along >>>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>> could be done as a feature enhancement? Thanks. >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/374b2ced/attachment-0002.html From mattdfong at gmail.com Fri Mar 20 06:06:35 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 20 Mar 2009 20:06:35 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> Message-ID: <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> Also, I would not be able to have a hang-up script do it, because in this scenario, the fifo consumer could hang-up at any time without any prior warning--otherwise, I could just transfer the fifo caller out before the fifo agent hangsup...but there is no prior warning :( --matt On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong wrote: > Hi Anthony, > I'm trying to use fifo in a different sense. Instead of using it for > inbound call queing, I'd like to use it for outbound call making. So > instead, my agents are "waiting" in the que, and once an outbound call is > connected, the "caller" will take an agent out of the que. > > So, in my case, the Fifo agent, would not be able to transfer the call > because it's an outbound call, and the phone number on the other side is > that of a non-employee. > > Fifo works a little smoother this way, because in reality, for outbound > call making to an agent, this is what's happening, not vica versa. How > difficult would this be to implement? Thanks. > > --matt > > 2009/3/20 Anthony Minessale > > The agent could transfer the caller to another extension. >> >> >> 2009/3/19 Matthew Fong >> >> Hi Anthony, >>> I installed the patch, but I don't think it accomplishes what I want. >>> >>> I want the opposite, I want the fifo caller to continue along with the >>> dialplan after the agent (consumer) is finished with servicing the call. >>> This might be useful in situations where there could be an IVR recording >>> customer satisfaction results of the agent servicing the call. As is, FS >>> ends the caller's channel after finishing up in the fifo (ie, agent >>> (consumer) disconnects or hangsup)--there should be life after s/he has been >>> serviced by an agent (preferably continuing on in the dialplan). >>> >>> If I'm confused and missing something obvious, please correct me... >>> Thanks >>> >>> --matt >>> >>> >>> >>> 2009/3/19 Anthony Minessale >>> >>> This is the patch >>>> >>>> http://jira.freeswitch.org/browse/MODAPP-237 >>>> >>>> it's not added yet. >>>> >>>> >>>> 2009/3/18 Matthew Fong >>>> >>>> I upgraded to >>>>> FreeSWITCH Version 1.0.trunk (12654M) >>>>> >>>>> but caller is still being hungup (and not continuing on with dialplan) >>>>> after agent disconnect with hangup_after_bridge=false >>>>> >>>>> Is there a separate patch I need to apply? Thanks. >>>>> >>>>> --matt >>>>> >>>>> >>>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>>>> >>>>>> Hi Anthony, thanks for the reply. >>>>>> I've searched thru jira, and didn't find anything when searching for >>>>>> fifo that was recently updated or related, except >>>>>> >>>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>>> >>>>>> and I'm not sure if this does what I need. Was this what you were >>>>>> referring to? Thanks. >>>>>> >>>>>> --matt >>>>>> >>>>>> 2009/3/17 Anthony Minessale >>>>>> >>>>>> there is a patch in jira that will implement this feature about to be >>>>>>> added >>>>>>> >>>>>>> >>>>>>> 2009/3/17 Matthew Fong >>>>>>> >>>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>>> (first timer posting here ;). >>>>>>>> >>>>>>>> I have a situation where it would be useful for a caller not to be >>>>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>>>> situation. It would be preferable if the caller channel went further along >>>>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>>> could be done as a feature enhancement? Thanks. >>>>>>>> >>>>>>>> --matt >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> >>>>>>> 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:213-799-1400 >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/a193b6b3/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 20 06:25:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Mar 2009 08:25:44 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903170637y5621540bn8ad5aca4e16f231b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> Message-ID: <191c3a030903200625k23edd429p38ec8b518ec97ee3@mail.gmail.com> Even though it's an outbound call if your agent uses his sip phone to blind transfer the caller "customer", The customer call will change the the routing state and hunt in your local dialplan just like it was an inbound call. That's how blind transfer was designed to work. If your agent is not using a sip phone, you can use bind_meta_app to make *N (where N = 0-9) to trigger a software blind transfer. 2009/3/20 Matthew Fong > Also, I would not be able to have a hang-up script do it, because in this > scenario, the fifo consumer could hang-up at any time without any prior > warning--otherwise, I could just transfer the fifo caller out before the > fifo agent hangsup...but there is no prior warning :( > --matt > > > On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong wrote: > >> Hi Anthony, >> I'm trying to use fifo in a different sense. Instead of using it for >> inbound call queing, I'd like to use it for outbound call making. So >> instead, my agents are "waiting" in the que, and once an outbound call is >> connected, the "caller" will take an agent out of the que. >> >> So, in my case, the Fifo agent, would not be able to transfer the call >> because it's an outbound call, and the phone number on the other side is >> that of a non-employee. >> >> Fifo works a little smoother this way, because in reality, for outbound >> call making to an agent, this is what's happening, not vica versa. How >> difficult would this be to implement? Thanks. >> >> --matt >> >> 2009/3/20 Anthony Minessale >> >> The agent could transfer the caller to another extension. >>> >>> >>> 2009/3/19 Matthew Fong >>> >>> Hi Anthony, >>>> I installed the patch, but I don't think it accomplishes what I want. >>>> >>>> I want the opposite, I want the fifo caller to continue along with the >>>> dialplan after the agent (consumer) is finished with servicing the call. >>>> This might be useful in situations where there could be an IVR recording >>>> customer satisfaction results of the agent servicing the call. As is, FS >>>> ends the caller's channel after finishing up in the fifo (ie, agent >>>> (consumer) disconnects or hangsup)--there should be life after s/he has been >>>> serviced by an agent (preferably continuing on in the dialplan). >>>> >>>> If I'm confused and missing something obvious, please correct me... >>>> Thanks >>>> >>>> --matt >>>> >>>> >>>> >>>> 2009/3/19 Anthony Minessale >>>> >>>> This is the patch >>>>> >>>>> http://jira.freeswitch.org/browse/MODAPP-237 >>>>> >>>>> it's not added yet. >>>>> >>>>> >>>>> 2009/3/18 Matthew Fong >>>>> >>>>> I upgraded to >>>>>> FreeSWITCH Version 1.0.trunk (12654M) >>>>>> >>>>>> but caller is still being hungup (and not continuing on with dialplan) >>>>>> after agent disconnect with hangup_after_bridge=false >>>>>> >>>>>> Is there a separate patch I need to apply? Thanks. >>>>>> >>>>>> --matt >>>>>> >>>>>> >>>>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>>>>> >>>>>>> Hi Anthony, thanks for the reply. >>>>>>> I've searched thru jira, and didn't find anything when searching for >>>>>>> fifo that was recently updated or related, except >>>>>>> >>>>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>>>> >>>>>>> and I'm not sure if this does what I need. Was this what you were >>>>>>> referring to? Thanks. >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> 2009/3/17 Anthony Minessale >>>>>>> >>>>>>> there is a patch in jira that will implement this feature about to be >>>>>>>> added >>>>>>>> >>>>>>>> >>>>>>>> 2009/3/17 Matthew Fong >>>>>>>> >>>>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>>>> (first timer posting here ;). >>>>>>>>> >>>>>>>>> I have a situation where it would be useful for a caller not to be >>>>>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>>>>> situation. It would be preferable if the caller channel went further along >>>>>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>>>> could be done as a feature enhancement? Thanks. >>>>>>>>> >>>>>>>>> --matt >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II >>>>>>>> >>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>> >>>>>>>> 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:213-799-1400 >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/0a3d4826/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 20 06:48:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Mar 2009 08:48:41 -0500 Subject: [Freeswitch-users] not hanging up In-Reply-To: <4ea6e8f20903191242vb56bcaetb7acd4853b215b0e@mail.gmail.com> References: <4ea6e8f20903191242vb56bcaetb7acd4853b215b0e@mail.gmail.com> Message-ID: <191c3a030903200648g9b10d78h5b81a79cbbeaa107@mail.gmail.com> It looks like interop issue with dialog matching between asterisk and freeswitch. Which version of asterisk is it? Which version of FreeSWITCH? You may want to provide a trace of the whole call starting with the invite. FS is having trouble identifying what call asterisk wants to cancel. 2009/3/19 Steven Ward > I have phones registered to a FS box, and an * box. There is a sip trunk > between the two boxes. > > A phone on my * (54321) calls a FS phone (12345); if I hang up the * phone > while it's still ringing, this is what I get on the sip trace on FS: > > ... > > 2009-03-19 15:05:40 [NOTICE] switch_ivr_originate.c:1692 > switch_ivr_originate() Ring Ready sofia/internal/12345 at 11.2.22.45! > recv 364 bytes from udp/[11.2.22.45]:5060 at 19:05:44.312950: > ------------------------------------------------------------------------ > CANCEL sip:12345 at b-pbx-sip-3.abc.xyz.netSIP/2.0 > Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport > From: "Steve" > >;tag=as25193d44 > To: > > > Call-ID: 0c0614d866a62841546cbf3340224682 at 11.2.22.45 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 328 bytes to udp/[11.2.22.45]:5060 at 19:05:44.313572: > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 11.2.22.45:5060;branch=z9hG4bK1c8fabcd;rport=5060 > From: "Steve" > >;tag=as25193d44 > To: > >;tag=c5Z8Q1e93p7KD > Call-ID: 0c0614d866a62841546cbf3340224682 at 11.2.22.45 > CSeq: 103 CANCEL > Content-Length: 0 > > -------------------------------------------------------- > > > The effect is that the FS keeps on ringing - it doesn't detect the hangup. > > > When I call from a FS phone (1000) to another FS phone (12345), and I hang > up the calling phone > while it's still ringing, this is what I get on the sip trace: > > ... > > send 425 bytes to udp/[11.2.56.106]:63054 at 19:15:29.737163: > ------------------------------------------------------------------------ > CANCEL sip:12345 at 11.2.56.106:63054;rinstance=64e968d7a1317bc3 SIP/2.0 > Via: SIP/2.0/UDP 11.2.22.46;rport;branch=z9hG4bKcraeFDFH4c68a > Max-Forwards: 69 > From: "Extension 1000" > >;tag=meK8yUgpgU2Zc > To: > Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 > CSeq: 112626727 CANCEL > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 427 bytes from udp/[11.2.56.106]:63054 at 19:15:29.838863: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a > Contact: > To: ;rinstance=64e968d7a1317bc3>;tag=db12c87a > From: "Extension 1000" > >;tag=meK8yUgpgU2Zc > Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 > CSeq: 112626727 CANCEL > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 376 bytes from udp/[11.2.56.106]:63054 at 19:15:29.839334: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 11.2.22.46;rport=5060;branch=z9hG4bKcraeFDFH4c68a > To: ;rinstance=64e968d7a1317bc3>;tag=db12c87a > From: "Extension 1000" > >;tag=meK8yUgpgU2Zc > Call-ID: 2593a17a-8f5d-122c-23b5-003018ae1862 > CSeq: 112626727 INVITE > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > > ... > > It works just fine. Any ideas? I'm not sure where to go with this. > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/f2b9e550/attachment-0002.html From mattdfong at gmail.com Fri Mar 20 07:27:31 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 20 Mar 2009 21:27:31 +0700 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <191c3a030903200625k23edd429p38ec8b518ec97ee3@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <191c3a030903170853x6659aa6ax3c15aec8e0dde023@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> <191c3a030903200625k23edd429p38ec8b518ec97ee3@mail.gmail.com> Message-ID: <4256bf830903200727g7259549dk8af17b4c583848d8@mail.gmail.com> Hi Anthony, Thanks for explaining blind transfer for me. The issue is that the fifo caller (my employee/agent, fifo in), gets hung-up on when the fifo consumer (an outside line to another party, fifo out) hangs up. I think this is because fifo was written under the assumption that the first in first out would always be a caller, and the agent would consume a caller. In my case, the roles are reversed, and there's no option to prevent the hangup of the caller. If the fifo caller (my employee/agent) could somehow know when a fifo consumer (my outside line to another party) was going to hangup, s/he could blind transfer out to save his/her connection from being hung-up, but unfortunately people don't always tell you before hand they are going to hangup. Right?!?!?! Thanks. --matt 2009/3/20 Anthony Minessale > Even though it's an outbound call if your agent uses his sip phone to blind > transfer the caller "customer", > The customer call will change the the routing state and hunt in your local > dialplan just like it was an inbound call. That's how blind transfer was > designed to work. > > If your agent is not using a sip phone, you can use bind_meta_app to make > *N (where N = 0-9) to trigger a software blind transfer. > > > 2009/3/20 Matthew Fong > > Also, I would not be able to have a hang-up script do it, because in this >> scenario, the fifo consumer could hang-up at any time without any prior >> warning--otherwise, I could just transfer the fifo caller out before the >> fifo agent hangsup...but there is no prior warning :( >> --matt >> >> >> On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong wrote: >> >>> Hi Anthony, >>> I'm trying to use fifo in a different sense. Instead of using it for >>> inbound call queing, I'd like to use it for outbound call making. So >>> instead, my agents are "waiting" in the que, and once an outbound call is >>> connected, the "caller" will take an agent out of the que. >>> >>> So, in my case, the Fifo agent, would not be able to transfer the call >>> because it's an outbound call, and the phone number on the other side is >>> that of a non-employee. >>> >>> Fifo works a little smoother this way, because in reality, for outbound >>> call making to an agent, this is what's happening, not vica versa. How >>> difficult would this be to implement? Thanks. >>> >>> --matt >>> >>> 2009/3/20 Anthony Minessale >>> >>> The agent could transfer the caller to another extension. >>>> >>>> >>>> 2009/3/19 Matthew Fong >>>> >>>> Hi Anthony, >>>>> I installed the patch, but I don't think it accomplishes what I want. >>>>> >>>>> I want the opposite, I want the fifo caller to continue along with the >>>>> dialplan after the agent (consumer) is finished with servicing the call. >>>>> This might be useful in situations where there could be an IVR recording >>>>> customer satisfaction results of the agent servicing the call. As is, FS >>>>> ends the caller's channel after finishing up in the fifo (ie, agent >>>>> (consumer) disconnects or hangsup)--there should be life after s/he has been >>>>> serviced by an agent (preferably continuing on in the dialplan). >>>>> >>>>> If I'm confused and missing something obvious, please correct me... >>>>> Thanks >>>>> >>>>> --matt >>>>> >>>>> >>>>> >>>>> 2009/3/19 Anthony Minessale >>>>> >>>>> This is the patch >>>>>> >>>>>> http://jira.freeswitch.org/browse/MODAPP-237 >>>>>> >>>>>> it's not added yet. >>>>>> >>>>>> >>>>>> 2009/3/18 Matthew Fong >>>>>> >>>>>> I upgraded to >>>>>>> FreeSWITCH Version 1.0.trunk (12654M) >>>>>>> >>>>>>> but caller is still being hungup (and not continuing on with >>>>>>> dialplan) after agent disconnect with hangup_after_bridge=false >>>>>>> >>>>>>> Is there a separate patch I need to apply? Thanks. >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> >>>>>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong wrote: >>>>>>> >>>>>>>> Hi Anthony, thanks for the reply. >>>>>>>> I've searched thru jira, and didn't find anything when searching for >>>>>>>> fifo that was recently updated or related, except >>>>>>>> >>>>>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>>>>> >>>>>>>> and I'm not sure if this does what I need. Was this what you were >>>>>>>> referring to? Thanks. >>>>>>>> >>>>>>>> --matt >>>>>>>> >>>>>>>> 2009/3/17 Anthony Minessale >>>>>>>> >>>>>>>> there is a patch in jira that will implement this feature about to >>>>>>>>> be added >>>>>>>>> >>>>>>>>> >>>>>>>>> 2009/3/17 Matthew Fong >>>>>>>>> >>>>>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>>>>> (first timer posting here ;). >>>>>>>>>> >>>>>>>>>> I have a situation where it would be useful for a caller not to be >>>>>>>>>> hungup, after finishing the "fifo in" execution (when the agent disconnects >>>>>>>>>> the call or the agent hangs-up). The caller is automatically hungup, in this >>>>>>>>>> situation. It would be preferable if the caller channel went further along >>>>>>>>>> the dial plan. I thought I might get lucky implementing this setting with >>>>>>>>>> hangup_after_bridge to false, but fifo does not utilize this variable. >>>>>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>>>>> could be done as a feature enhancement? Thanks. >>>>>>>>>> >>>>>>>>>> --matt >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Freeswitch-users mailing list >>>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Anthony Minessale II >>>>>>>>> >>>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>>> >>>>>>>>> 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:213-799-1400 >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/7a982ed3/attachment-0002.html From Mark.Tabron at rnid-typetalk.org.uk Fri Mar 20 09:09:00 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Fri, 20 Mar 2009 16:09:00 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE76536F@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765375@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903161015v7f1e6a9ax2ef7b01c830367ab@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE765376@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765377@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903170848h250d7128rb85f4fd911f2d342@mail.gmail.com><11C1F78E88546B4387E9CC0603051CFE76537D@tt-mail.RNID.TYPETALK.LOCAL><49C12CD8.7020203@gmx.net><11C1F78E88546B4387E9CC0603051CFE76537E@tt-mail.RNID.TYPETALK.LOCAL><87f2f3b90903190911s6082877at1231c27f6a86506@mail.gmail.com> <11C1F78E88546B4387E9CC0603051CFE765381@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765382@tt-mail.RNID.TYPETALK.LOCAL> Installed libpri but I'm stuck on what entries to put in openzap.conf.xml, here's how I have the span setup at the moment: Node and Switch type aren't documented for libpri from what I can tell - I know the former is either CPE or NET, though, I'm unsure what other values can be used for switch type. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 19 March 2009 16:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Thanks, found an install guide on the FS Wiki for libpri - will get the server cloned then install and test. Shall report back. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 19 March 2009 16:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron wrote: > So the second issue is possibly known - really could do with a fix or a > workaround for this as we plan to use E1's for all incoming traffic. > > Can anyone shed light on the first problem (extension rings for a > fraction of a second then hangs up) I mentioned below, or is that > possibly part of the same issue? I have experienced this before but I believe it was resolved by having the telco switch protocol dialects which is probably not an option for you. I think your best bet is to use ozmod_libpri and see if the issue is still present. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. ------------------------------------------------------------------------ -------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). ------------------------------------------------------------------------ -------- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From stkn at freeswitch.org Fri Mar 20 11:29:05 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Fri, 20 Mar 2009 19:29:05 +0100 Subject: [Freeswitch-users] Problem dialing out via E1 In-Reply-To: <11C1F78E88546B4387E9CC0603051CFE765382@tt-mail.RNID.TYPETALK.LOCAL> References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765381@tt-mail.RNID.TYPETALK.LOCAL> <11C1F78E88546B4387E9CC0603051CFE765382@tt-mail.RNID.TYPETALK.LOCAL> Message-ID: <200903201929.05829.stkn@freeswitch.org> Am Friday 20 March 2009 schrieb Mark Tabron: > Installed libpri but I'm stuck on what entries to put in > openzap.conf.xml, here's how I have the span setup at the moment: > > > > > > > > > > > Node and Switch type aren't documented for libpri from what I can tell - > I know the former is either CPE or NET, though, I'm unsure what other > values can be used for switch type. > The value for switch is invalid, it's going to fall back to dms100 with that one set. Valid settings are: ni1, ni2, dms100, euroisdn, lucent5e, att4ess, gr303eoc and gr303tmc and "euroisdn" is the one you'll want for a E1 line. Another setting you may need is: stkn -- ------------------------------------------------------------------------------- Stefan Knoblich | axsentis GmbH | Web: http://www.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | http://oss.axsentis.de/ Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From anthony.minessale at gmail.com Fri Mar 20 12:04:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Mar 2009 14:04:04 -0500 Subject: [Freeswitch-users] Fifo feature request -- no caller disconnect after agent hangup In-Reply-To: <4256bf830903200727g7259549dk8af17b4c583848d8@mail.gmail.com> References: <4256bf830903170558k70986793r8371fd6de505617b@mail.gmail.com> <4256bf830903172139q574e38daxbf3d54429ad691ce@mail.gmail.com> <4256bf830903180859g38e6e94dp589c89fa83393464@mail.gmail.com> <191c3a030903181200n576253cch7de94ec096ed48e7@mail.gmail.com> <4256bf830903192046kdc0868jcd0e1097fbb61aa4@mail.gmail.com> <191c3a030903200504v34cd779dr24d1ce7532bbf8c@mail.gmail.com> <4256bf830903200604w3ec73f16v7f85f6e973a5e1a0@mail.gmail.com> <4256bf830903200606i9ebcbc9o9e22b2e524efa185@mail.gmail.com> <191c3a030903200625k23edd429p38ec8b518ec97ee3@mail.gmail.com> <4256bf830903200727g7259549dk8af17b4c583848d8@mail.gmail.com> Message-ID: <191c3a030903201204l5dc16ad1pee4f4635e1addef7@mail.gmail.com> I added a feature to latest trunk where you can set the variable transfer_after_bridge=xyz where xyz is an extension you want to transfer to when the call ends assuming it's not hungup yet. if you set this in your dialplan before entering fifo it will allow you to be transfered to the desired ext when you are done. if you need to specify a dialplan and context you cana add that too [:][:] 2009/3/20 Matthew Fong > Hi Anthony, > Thanks for explaining blind transfer for me. > > The issue is that the fifo caller (my employee/agent, fifo in), gets > hung-up on when the fifo consumer (an outside line to another party, fifo > out) hangs up. I think this is because fifo was written under the assumption > that the first in first out would always be a caller, and the agent would > consume a caller. > > In my case, the roles are reversed, and there's no option to prevent the > hangup of the caller. > > If the fifo caller (my employee/agent) could somehow know when a fifo > consumer (my outside line to another party) was going to hangup, s/he could > blind transfer out to save his/her connection from being hung-up, but > unfortunately people don't always tell you before hand they are going to > hangup. > > Right?!?!?! Thanks. > > --matt > > 2009/3/20 Anthony Minessale > >> Even though it's an outbound call if your agent uses his sip phone to >> blind transfer the caller "customer", >> The customer call will change the the routing state and hunt in your local >> dialplan just like it was an inbound call. That's how blind transfer was >> designed to work. >> >> If your agent is not using a sip phone, you can use bind_meta_app to make >> *N (where N = 0-9) to trigger a software blind transfer. >> >> >> 2009/3/20 Matthew Fong >> >> Also, I would not be able to have a hang-up script do it, because in this >>> scenario, the fifo consumer could hang-up at any time without any prior >>> warning--otherwise, I could just transfer the fifo caller out before the >>> fifo agent hangsup...but there is no prior warning :( >>> --matt >>> >>> >>> On Fri, Mar 20, 2009 at 8:04 PM, Matthew Fong wrote: >>> >>>> Hi Anthony, >>>> I'm trying to use fifo in a different sense. Instead of using it for >>>> inbound call queing, I'd like to use it for outbound call making. So >>>> instead, my agents are "waiting" in the que, and once an outbound call is >>>> connected, the "caller" will take an agent out of the que. >>>> >>>> So, in my case, the Fifo agent, would not be able to transfer the call >>>> because it's an outbound call, and the phone number on the other side is >>>> that of a non-employee. >>>> >>>> Fifo works a little smoother this way, because in reality, for outbound >>>> call making to an agent, this is what's happening, not vica versa. How >>>> difficult would this be to implement? Thanks. >>>> >>>> --matt >>>> >>>> 2009/3/20 Anthony Minessale >>>> >>>> The agent could transfer the caller to another extension. >>>>> >>>>> >>>>> 2009/3/19 Matthew Fong >>>>> >>>>> Hi Anthony, >>>>>> I installed the patch, but I don't think it accomplishes what I want. >>>>>> >>>>>> I want the opposite, I want the fifo caller to continue along with the >>>>>> dialplan after the agent (consumer) is finished with servicing the call. >>>>>> This might be useful in situations where there could be an IVR recording >>>>>> customer satisfaction results of the agent servicing the call. As is, FS >>>>>> ends the caller's channel after finishing up in the fifo (ie, agent >>>>>> (consumer) disconnects or hangsup)--there should be life after s/he has been >>>>>> serviced by an agent (preferably continuing on in the dialplan). >>>>>> >>>>>> If I'm confused and missing something obvious, please correct me... >>>>>> Thanks >>>>>> >>>>>> --matt >>>>>> >>>>>> >>>>>> >>>>>> 2009/3/19 Anthony Minessale >>>>>> >>>>>> This is the patch >>>>>>> >>>>>>> http://jira.freeswitch.org/browse/MODAPP-237 >>>>>>> >>>>>>> it's not added yet. >>>>>>> >>>>>>> >>>>>>> 2009/3/18 Matthew Fong >>>>>>> >>>>>>> I upgraded to >>>>>>>> FreeSWITCH Version 1.0.trunk (12654M) >>>>>>>> >>>>>>>> but caller is still being hungup (and not continuing on with >>>>>>>> dialplan) after agent disconnect with hangup_after_bridge=false >>>>>>>> >>>>>>>> Is there a separate patch I need to apply? Thanks. >>>>>>>> >>>>>>>> --matt >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong >>>>>>> > wrote: >>>>>>>> >>>>>>>>> Hi Anthony, thanks for the reply. >>>>>>>>> I've searched thru jira, and didn't find anything when searching >>>>>>>>> for fifo that was recently updated or related, except >>>>>>>>> >>>>>>>>> http://jira.freeswitch.org/browse/MODAPP-189 >>>>>>>>> >>>>>>>>> and I'm not sure if this does what I need. Was this what you were >>>>>>>>> referring to? Thanks. >>>>>>>>> >>>>>>>>> --matt >>>>>>>>> >>>>>>>>> 2009/3/17 Anthony Minessale >>>>>>>>> >>>>>>>>> there is a patch in jira that will implement this feature about to >>>>>>>>>> be added >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2009/3/17 Matthew Fong >>>>>>>>>> >>>>>>>>>>> I apologize if this is a double post to -dev. I'm not sure why I >>>>>>>>>>> don't see my message appearing, so I'm going to try again in the -user list >>>>>>>>>>> (first timer posting here ;). >>>>>>>>>>> >>>>>>>>>>> I have a situation where it would be useful for a caller not to >>>>>>>>>>> be hungup, after finishing the "fifo in" execution (when the agent >>>>>>>>>>> disconnects the call or the agent hangs-up). The caller is automatically >>>>>>>>>>> hungup, in this situation. It would be preferable if the caller channel went >>>>>>>>>>> further along the dial plan. I thought I might get lucky implementing this >>>>>>>>>>> setting with hangup_after_bridge to false, but fifo does not utilize this >>>>>>>>>>> variable. >>>>>>>>>>> I tried looking thru the mod_fifo.c source, but my c skills are >>>>>>>>>>> minimal. I also tried executing fifo in a lua app and >>>>>>>>>>> setting setAutoHangup(false), but that also did not work. Any chance this >>>>>>>>>>> could be done as a feature enhancement? Thanks. >>>>>>>>>>> >>>>>>>>>>> --matt >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> Freeswitch-users mailing list >>>>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Anthony Minessale II >>>>>>>>>> >>>>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>>>> >>>>>>>>>> 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:213-799-1400 >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Freeswitch-users mailing list >>>>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> >>>>>>> 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:213-799-1400 >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/7f6214fa/attachment-0002.html From switchserver at gmail.com Fri Mar 20 13:14:02 2009 From: switchserver at gmail.com (HarryK) Date: Fri, 20 Mar 2009 13:14:02 -0700 (PDT) Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> Message-ID: <22628039.post@talk.nabble.com> Yes that page was perfect, ty. (Cepstral 5.1) One thing. Is there a way to set the default voice used? I have 2 voices installed and it goes to the first one by default. It also refers to this voice as "David" even though its Allison. I'd like it to always use the second voice for RSS. There is nothing in the configs, I figured maybe it was yet to be documented. I hope Cepstral is more stable in the future, I havent used it much so noticed no problems yet. Thanks again Carlos Talbot wrote: > > I wrote this wiki page a while back. Did it help? > > http://wiki.freeswitch.org/wiki/Mod_rss > > > On Thu, Mar 19, 2009 at 2:41 AM, HarryK wrote: > >> >> I have Cepstral working. >> >> Can someone please tell me how to go about having it read RSS feeds? I >> can >> have the dialplan direct it np. But I really dont have a clue how to >> point >> it at an RSS. Any help would be great, ddint find anything in the wiki. >> >> >> -- >> View this message in context: >> http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22594923.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Cepstral-and-RSS-feeds-tp22594923p22628039.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gkuri at ieee.org Fri Mar 20 14:28:35 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 20 Mar 2009 14:28:35 -0700 Subject: [Freeswitch-users] PCMU fallback for T.38 Message-ID: <49C40A83.1050003@ieee.org> hey folks, I'm trying to configure PCMU fallback for T.38. The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with G729 and PCMU in the sdp. the INVITE to the provider includes G729 and PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... m=audio 16458 RTP/AVP 18 0 100 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 once the FAX tone is detected on the PSTN side, FS receives a T.38 re-INVITE from the provider and FS sends back a 488/Not Acceptable (proxy_media=false). at that point the provider than attempts fallback to PCMU with another reINVITE ... m=audio 16816 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 immediately after the PCMU reINVITE, FS closes the channel and the text below is in the FS logs. given the SPA-2102 included PCMU in the original INVITE, even though it was the second preferred codec, shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by the provider? It seems like it's not passing the PCMU Re-INVITE back to the endpoint (SPA-2102), since it originally negotiated G729 with the SPA2102 as that was the 1st codec in the sdp, but trying to transcode between the two (G729 and PCMU)? 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 PCMU/8000 20 ms 160 samples 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1. 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing Reinvite 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 entering state [completed] 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! Thanks, Gabe From brian at freeswitch.org Fri Mar 20 14:35:14 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Mar 2009 16:35:14 -0500 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49C40A83.1050003@ieee.org> References: <49C40A83.1050003@ieee.org> Message-ID: <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> Are you on SVN trunk 12694? /b On Mar 20, 2009, at 4:28 PM, Gabriel Kuri wrote: > hey folks, I'm trying to configure PCMU fallback for T.38. > > The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with > G729 and PCMU in the sdp. the INVITE to the provider includes G729 and > PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... > > m=audio 16458 RTP/AVP 18 0 100 101 > a=rtpmap:18 G729/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > once the FAX tone is detected on the PSTN side, FS receives a T.38 > re-INVITE from the provider and FS sends back a 488/Not Acceptable > (proxy_media=false). at that point the provider than attempts fallback > to PCMU with another reINVITE ... > > m=audio 16816 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > immediately after the PCMU reINVITE, FS closes the channel and the > text > below is in the FS logs. given the SPA-2102 included PCMU in the > original INVITE, even though it was the second preferred codec, > shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by > the provider? It seems like it's not passing the PCMU Re-INVITE back > to > the endpoint (SPA-2102), since it originally negotiated G729 with the > SPA2102 as that was the 1st codec in the sdp, but trying to transcode > between the two (G729 and PCMU)? > > > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 > sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 > sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 > sofia_glue_negotiate_sdp() > Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 > sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 > sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 > sofia_glue_tech_set_codec() Set Codec > sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 PCMU/8000 20 ms 160 > samples > 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 > sofia_glue_activate_rtp() > Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 > . > 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing Reinvite > 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 entering state > [completed] > 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 > switch_core_session_write_frame() > sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 receive message > [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This > codec > is only usable in passthrough mode! > 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 > switch_core_session_write_frame() Codec G.729 decoder error! > > > Thanks, > > Gabe > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gkuri at ieee.org Fri Mar 20 14:41:47 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 20 Mar 2009 14:41:47 -0700 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> References: <49C40A83.1050003@ieee.org> <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> Message-ID: <49C40D9B.6000803@ieee.org> err, no, I tried upgrading from r11000 to r12669 yesterday, but starting seeing crashing, so I have a jira open. currently I'm back on r11000. http://jira.freeswitch.org/browse/FSCORE-338 Gabe Brian West wrote: > Are you on SVN trunk 12694? > > /b > > On Mar 20, 2009, at 4:28 PM, Gabriel Kuri wrote: > >> hey folks, I'm trying to configure PCMU fallback for T.38. >> >> The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with >> G729 and PCMU in the sdp. the INVITE to the provider includes G729 and >> PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... >> >> m=audio 16458 RTP/AVP 18 0 100 101 >> a=rtpmap:18 G729/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:100 NSE/8000 >> a=fmtp:100 192-193 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> once the FAX tone is detected on the PSTN side, FS receives a T.38 >> re-INVITE from the provider and FS sends back a 488/Not Acceptable >> (proxy_media=false). at that point the provider than attempts fallback >> to PCMU with another reINVITE ... >> >> m=audio 16816 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> immediately after the PCMU reINVITE, FS closes the channel and the >> text >> below is in the FS logs. given the SPA-2102 included PCMU in the >> original INVITE, even though it was the second preferred codec, >> shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by >> the provider? It seems like it's not passing the PCMU Re-INVITE back >> to >> the endpoint (SPA-2102), since it originally negotiated G729 with the >> SPA2102 as that was the 1st codec in the sdp, but trying to transcode >> between the two (G729 and PCMU)? >> >> >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 >> sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 >> sofia_glue_negotiate_sdp() >> Set 2833 dtmf payload to 101 >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 >> sofia_glue_negotiate_sdp() >> Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 >> sofia_glue_negotiate_sdp() >> Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 >> sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 >> sofia_glue_tech_set_codec() Set Codec >> sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 PCMU/8000 20 ms 160 >> samples >> 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 >> sofia_glue_activate_rtp() >> Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 >> . >> 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() >> Processing Reinvite >> 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() >> Channel sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 entering state >> [completed] >> 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 >> switch_core_session_write_frame() >> sofia/cedarwireless.net/1XXXXXXXXXX at 1.1.1.1 receive message >> [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] >> 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This >> codec >> is only usable in passthrough mode! >> 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 >> switch_core_session_write_frame() Codec G.729 decoder error! >> >> >> Thanks, >> >> Gabe >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri Mar 20 14:48:43 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Mar 2009 16:48:43 -0500 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49C40D9B.6000803@ieee.org> References: <49C40A83.1050003@ieee.org> <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> <49C40D9B.6000803@ieee.org> Message-ID: <59121C75-9F16-4794-AA91-9F71DA3B6E7D@freeswitch.org> Make current and try again... I haven't seen this crash you have seen... if you can run sippcapdump and get the packets that would help also. Thanks, /b On Mar 20, 2009, at 4:41 PM, Gabriel Kuri wrote: > err, no, I tried upgrading from r11000 to r12669 yesterday, but > starting > seeing crashing, so I have a jira open. currently I'm back on r11000. > > http://jira.freeswitch.org/browse/FSCORE-338 > > Gabe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/814dd93d/attachment-0002.html From gkuri at ieee.org Fri Mar 20 16:04:07 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 20 Mar 2009 16:04:07 -0700 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <59121C75-9F16-4794-AA91-9F71DA3B6E7D@freeswitch.org> References: <49C40A83.1050003@ieee.org> <28B1787F-32AF-45CD-9FA2-84937910FDC8@freeswitch.org> <49C40D9B.6000803@ieee.org> <59121C75-9F16-4794-AA91-9F71DA3B6E7D@freeswitch.org> Message-ID: <49C420E7.0@ieee.org> OK, I'll give it a try and report back. Gabe Brian West wrote: > Make current and try again... I haven't seen this crash you have seen... > if you can run sippcapdump and get the packets that would help also. > > Thanks, > /b > > > On Mar 20, 2009, at 4:41 PM, Gabriel Kuri wrote: > >> err, no, I tried upgrading from r11000 to r12669 yesterday, but starting >> seeing crashing, so I have a jira open. currently I'm back on r11000. >> >> http://jira.freeswitch.org/browse/FSCORE-338 >> >> Gabe > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave at 3c.co.uk Fri Mar 20 18:53:46 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 20 Mar 2009 19:53:46 -0600 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> Message-ID: <49C448AA.6070207@3c.co.uk> In the meantime, you can work around this by using the swift executable to turn text in to WAV files, and then just play them back. Works fine for short(ish) texts - there might be a bit of a delay if you wanted the thing to read back War and Peace. --Dave > http://jira.freeswitch.org/browse/MODASRTTS-11 > > Might wanna know about that issue also :) > > /b > > On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote: > >> I wrote this wiki page a while back. Did it help? >> >> http://wiki.freeswitch.org/wiki/Mod_rss > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/25851bbb/attachment-0002.html From brian at freeswitch.org Fri Mar 20 19:02:40 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Mar 2009 21:02:40 -0500 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <49C448AA.6070207@3c.co.uk> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> <49C448AA.6070207@3c.co.uk> Message-ID: <1C27B16A-898C-4DAA-9B1C-3E7DB4A15F22@freeswitch.org> Oxymoron!?!?! :) /b On Mar 20, 2009, at 8:53 PM, David Knell wrote: > War and Peace. From jason at jasonjgw.net Fri Mar 20 19:10:28 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 21 Mar 2009 13:10:28 +1100 Subject: [Freeswitch-users] proper way to clean source tree after build? Message-ID: <20090321021028.GA5399@jdc.jasonjgw.net> I'm building under Debian. To clean the source tree after building, I've tried make clean debuild clean but what often happens is that after updating to a later svn revision and then trying to build, I get various errors during the compilation process that disappear if I make a clean checkout from the repository and run the build again. I have also found that using svn export to reproduce the tree in another directory, then building from that directory, doesn't always solve the problem. Is there a better way to clean up the source tree after updating it, to avoid these issues? I assume there are extraneous modifications or files that simply aren't being eliminated by the "clean" makefile targets. From dave at 3c.co.uk Fri Mar 20 19:34:12 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 20 Mar 2009 20:34:12 -0600 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <1C27B16A-898C-4DAA-9B1C-3E7DB4A15F22@freeswitch.org> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> <49C448AA.6070207@3c.co.uk> <1C27B16A-898C-4DAA-9B1C-3E7DB4A15F22@freeswitch.org> Message-ID: <49C45224.6010106@3c.co.uk> It's a book, Brian, and a long one at that, often printed in black and white - oops, there I go again ;-) I once referred to "oxymoronic hip-hop culture", only to be firmly told by a listener that there was nothing moronic about hip-hop.. Cheers -- Dave > Oxymoron!?!?! :) > > /b > > On Mar 20, 2009, at 8:53 PM, David Knell wrote: > > >> War and Peace. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/ed3c15b8/attachment-0002.html From steveu at coppice.org Fri Mar 20 19:46:32 2009 From: steveu at coppice.org (Steve Underwood) Date: Sat, 21 Mar 2009 10:46:32 +0800 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49C40A83.1050003@ieee.org> References: <49C40A83.1050003@ieee.org> Message-ID: <49C45508.402@coppice.org> Gabriel Kuri wrote: > once the FAX tone is detected on the PSTN side, FS receives a T.38 > re-INVITE from the provider and FS sends back a 488/Not Acceptable > (proxy_media=false). at that point the provider than attempts fallback > to PCMU with another reINVITE ... > This part is interesting, and the subject of a discussion we had recently. A number of systems try that second re-invite after a 488, but the SIP specs say the call is pretty much dead after the 488 message is exchanged. Are they just hoping that maybe the other end will be non-compliant enough to keep the call alive, and recover its media mode, or haven't they read the specs? Steve From pablosaro at gmail.com Fri Mar 20 19:54:03 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Fri, 20 Mar 2009 23:54:03 -0300 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <153785972B6D42C99E8CAF7DE2F1A145@wsandy> References: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> <153785972B6D42C99E8CAF7DE2F1A145@wsandy> Message-ID: <247f8100903201954gcc81d83lf199c6a9b26463b4@mail.gmail.com> Hi Andy, Did you get Matheu's? > if you are behind NAT it is possible that your router "forgot" the mapping betweeen FS and your > provider, try adding to your gateway. Pablo 2009/3/20 Andy Ayers : > Thanks Brian, > > I've upgraded to 1.0.3 and things seem a little better but I'm still loosing > the gateway connection intermittently. I rebuilt the config on upgrade, is > there any possibility I've missed something? Is there a keep-alive setting > for a gateway or a re-connect after x or something. > > Many thanks for your help. > Andy > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian > West > Sent: 18 March 2009 14:08 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Losing Gateway registration > > Upgrade to 1.03 or SVN Trunk > /b > On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: > > Hi, > > I've recently ugrade to version 1.02 of freeswitch and am having some > problems with my gateway registrations. The gateway successfully registers > with my voip provider when freeswitch first starts but if left running it > seems to loose it's connection to my voip provider. I can get it to > reconnect with a sofia restart. I'm using the same provider and user account > as with the old version of the software. Can you suggest any reaosn why this > may be happening and how I can prevent it? > > Many thanks > Andy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From pablosaro at gmail.com Fri Mar 20 20:15:58 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Sat, 21 Mar 2009 00:15:58 -0300 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <49C34E54.9000000@czerpak.eu> References: <49C28D4F.4040307@czerpak.eu> <2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com> <49C29AF2.2000703@czerpak.eu> <49C34E54.9000000@czerpak.eu> Message-ID: <247f8100903202015g6befe872v73fe4a53585d6a60@mail.gmail.com> I am not a Cisco expert, but as far as I know packetization period is configurable in Cisco. When you specify the codec for a dial peer, you can set up the ptime value in milliseconds. This is an optional argument in the Cisco command line and what happens when providers don't specify it is that your end assumes a convenient or default ptime value, that probably don't match with far end default... A solution would be to figure out what is the convenient value for an individual provider and set up FS to match it, or suggest your provider to specify a ptime in the first request. By the way, Cisco supports 10, 20, 30, 40, 50 and 60 as ptime values for codecs G.729, G.729A, G.729B and G.729AB. Pablo 2009/3/20 ?ukasz Czerpak : > Brian West pisze: >> Well you can't have ptime 60 one way and 20 the other it just won't >> work. ?Also I can't even think that this illegal codec was even tested >> at 60ms... Try it with ulaw and see what it does. ?or only allow >> G729 at 60i and see what it does. >> > > I made some tests with ulaw with success. There is no problem with ptime > negotiation. > > Will the g729 codec be fully (not passthrough) supported in FreeSWITCH? > > regards, > > -- > ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mszlazak at aol.com Fri Mar 20 23:22:10 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 21 Mar 2009 02:22:10 -0400 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <49C448AA.6070207@3c.co.uk> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> <49C448AA.6070207@3c.co.uk> Message-ID: <8CB7804802401C1-1388-2309@webmail-mf01.sysops.aol.com> This product has much better sounding TTS than Cepstral: http://www.neospeech.com/Default.aspx Maybe you can use Dave's suggestion and make WAV recordings from their demo text input box and then just clip off the initial portion. Mark. -----Original Message----- From: David Knell To: freeswitch-users at lists.freeswitch.org Sent: Fri, 20 Mar 2009 6:53 pm Subject: Re: [Freeswitch-users] Cepstral and RSS feeds In the meantime, you can work around this by using the swift executable to turn text in to WAV files, and then just play them back.? Works fine for short(ish) texts - there might be a bit of a delay if you wanted the thing to read back War and Peace. --Dave http://jira.freeswitch.org/browse/MODASRTTS-11 Might wanna know about that issue also :) /b On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote: I wrote this wiki page a while back. Did it help? http://wiki.freeswitch.org/wiki/Mod_rss _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090321/45f047da/attachment-0002.html From anthony.minessale at gmail.com Sat Mar 21 05:56:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Mar 2009 07:56:02 -0500 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <8CB7804802401C1-1388-2309@webmail-mf01.sysops.aol.com> References: <22594923.post@talk.nabble.com> <5800526b0903192202k441c11dcl505cb3474f3dbb65@mail.gmail.com> <49C448AA.6070207@3c.co.uk> <8CB7804802401C1-1388-2309@webmail-mf01.sysops.aol.com> Message-ID: <191c3a030903210556j3611b0edh172a83ed0789acd1@mail.gmail.com> the rss app takes a space sep string of 3 optional args to control the voice 2009/3/21 > This product has much better sounding TTS than Cepstral: > > http://www.neospeech.com/Default.aspx > > Maybe you can use Dave's suggestion and make WAV recordings from their demo > text input box and then just clip off the initial portion. > > Mark. > > > -----Original Message----- > From: David Knell > To: freeswitch-users at lists.freeswitch.org > Sent: Fri, 20 Mar 2009 6:53 pm > Subject: Re: [Freeswitch-users] Cepstral and RSS feeds > > In the meantime, you can work around this by using the swift executable > to turn text in to WAV files, and then just play them back. Works fine for > short(ish) texts - there might be a bit of a delay if you wanted the thing > to > read back War and Peace. > > --Dave > > http://jira.freeswitch.org/browse/MODASRTTS-11 > Might wanna know about that issue also :) > > /b > > On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote: > > I wrote this wiki page a while back. Did it help? > > http://wiki.freeswitch.org/wiki/Mod_rss > > > ------------------------------ > > > _______________________________________________ > > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > *A Bad Credit Score is 600 or Below. See yours in just 2 easy steps! > * > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090321/09228fd0/attachment-0002.html From alhakeem at gmail.com Sat Mar 21 11:46:54 2009 From: alhakeem at gmail.com (Abdul Hakeem) Date: Sat, 21 Mar 2009 18:46:54 -0000 Subject: [Freeswitch-users] ptime problem with provider (Cisco hardware) In-Reply-To: <247f8100903202015g6befe872v73fe4a53585d6a60@mail.gmail.com> References: <49C28D4F.4040307@czerpak.eu><2d9149cd0903191159x6d97b2egc8a3ecd1ec63cdb0@mail.gmail.com><49C29AF2.2000703@czerpak.eu><49C34E54.9000000@czerpak.eu> <247f8100903202015g6befe872v73fe4a53585d6a60@mail.gmail.com> Message-ID: The pvalue of the originator overrides whatever you might have configured on Cisco GW. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Pablo Hernan Saro Sent: 21 March 2009 03:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] ptime problem with provider (Cisco hardware) I am not a Cisco expert, but as far as I know packetization period is configurable in Cisco. When you specify the codec for a dial peer, you can set up the ptime value in milliseconds. This is an optional argument in the Cisco command line and what happens when providers don't specify it is that your end assumes a convenient or default ptime value, that probably don't match with far end default... A solution would be to figure out what is the convenient value for an individual provider and set up FS to match it, or suggest your provider to specify a ptime in the first request. By the way, Cisco supports 10, 20, 30, 40, 50 and 60 as ptime values for codecs G.729, G.729A, G.729B and G.729AB. Pablo 2009/3/20 ?ukasz Czerpak : > Brian West pisze: >> Well you can't have ptime 60 one way and 20 the other it just won't >> work. ?Also I can't even think that this illegal codec was even >> tested at 60ms... Try it with ulaw and see what it does. ?or only >> allow G729 at 60i and see what it does. >> > > I made some tests with ulaw with success. There is no problem with > ptime negotiation. > > Will the g729 codec be fully (not passthrough) supported in FreeSWITCH? > > regards, > > -- > ?ukasz Czerpak | PGP: 0x532D8E1B [subkeys.pgp.net] > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From zhaoxxqq at 163.com Sun Mar 22 03:22:19 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Sun, 22 Mar 2009 18:22:19 +0800 Subject: [Freeswitch-users] Problem about always Double accept DTMFs Message-ID: <200903221822142515830@163.com> I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS to PSTN with DID numbers. For inband I connect it to FS?s demo_ivr. When I press the key, the FS accept always DOUBLE of key. The debug information like below. 2009-03-22 17:50:26 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 2009-03-22 17:50:26 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-03-22 17:50:26 [DEBUG] switch_ivr_menu.c:308 play_and_collect() waiting for 3/4 digits t/o 2000 2009-03-22 17:50:26 [DEBUG] sofia.c:3753 sofia_handle_sip_i_info() INFO DTMF(1) 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:353 play_and_collect() digits '11' 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:523 switch_ivr_menu_execute() IVR menu 'jtq_greating' caught invalid input '11' 2009-03-22 17:50:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-03-22 17:50:28 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/external/13323015 Can any friend can help me? Zhao Xiaoqiang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090322/5c47549a/attachment-0002.html From dave at 3c.co.uk Sun Mar 22 06:02:56 2009 From: dave at 3c.co.uk (David Knell) Date: Sun, 22 Mar 2009 07:02:56 -0600 Subject: [Freeswitch-users] Problem about always Double accept DTMFs In-Reply-To: <200903221822142515830@163.com> References: <200903221822142515830@163.com> Message-ID: <49C63700.8010104@3c.co.uk> Hi - It looks like you're getting digits both in the RTP stream and as SIP INFO. Try adding to the SIP profile you're using for inbound calls. --Dave > I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS > to PSTN with DID numbers. For inband I connect it to FS's demo_ivr. > When I press the key, the FS accept always DOUBLE of key. The debug > information like below. > > 2009-03-22 17:50:26 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 > 2009-03-22 17:50:26 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file > 2009-03-22 17:50:26 [DEBUG] switch_ivr_menu.c:308 play_and_collect() waiting for 3/4 digits t/o 2000 > 2009-03-22 17:50:26 [DEBUG] sofia.c:3753 sofia_handle_sip_i_info() INFO DTMF(1) > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:353 play_and_collect() digits '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:523 switch_ivr_menu_execute() IVR menu 'jtq_greating' caught invalid input '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-03-22 17:50:28 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/external/13323015 > > > Can any friend can help me? > > Zhao Xiaoqiang > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090322/45822ad7/attachment-0002.html From herman.griffin at gmail.com Fri Mar 20 22:33:34 2009 From: herman.griffin at gmail.com (Herman Griffin) Date: Fri, 20 Mar 2009 22:33:34 -0700 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <247f8100903201954gcc81d83lf199c6a9b26463b4@mail.gmail.com> References: <22992920-96B9-4AB7-A843-1995B933AE74@freeswitch.org> <153785972B6D42C99E8CAF7DE2F1A145@wsandy> <247f8100903201954gcc81d83lf199c6a9b26463b4@mail.gmail.com> Message-ID: <4d6f26b0903202233v72046523s19b2dace2c8cce55@mail.gmail.com> I was having the same problem (i think). It resolved it by adding gateways to the /usr/local/freeswitch/conf/directory/default.xml file instead of the /usr/local/freeswitch/sip_profiles/external/ directory. It may have been a total coincidence because I was so frustrated with the problem that I start from scratch. I backed up my conf directory and started configuring based on the example.com.xml template. However, it worked. I have had a loss of registration problem since then. ./herman griffin On Fri, Mar 20, 2009 at 7:54 PM, Pablo Hernan Saro wrote: > Hi Andy, > > Did you get Matheu's? > > > if you are behind NAT it is possible that your router "forgot" the > mapping betweeen FS and your > provider, try adding value="30" /> to your gateway. > > Pablo > > 2009/3/20 Andy Ayers : > > Thanks Brian, > > > > I've upgraded to 1.0.3 and things seem a little better but I'm still > loosing > > the gateway connection intermittently. I rebuilt the config on upgrade, > is > > there any possibility I've missed something? Is there a keep-alive > setting > > for a gateway or a re-connect after x or something. > > > > Many thanks for your help. > > Andy > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian > > West > > Sent: 18 March 2009 14:08 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Losing Gateway registration > > > > Upgrade to 1.03 or SVN Trunk > > /b > > On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: > > > > Hi, > > > > I've recently ugrade to version 1.02 of freeswitch and am having some > > problems with my gateway registrations. The gateway successfully > registers > > with my voip provider when freeswitch first starts but if left running it > > seems to loose it's connection to my voip provider. I can get it to > > reconnect with a sofia restart. I'm using the same provider and user > account > > as with the old version of the software. Can you suggest any reaosn why > this > > may be happening and how I can prevent it? > > > > Many thanks > > Andy > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090320/7ff465e6/attachment-0002.html From solko at gcdf.pl Sat Mar 21 01:29:38 2009 From: solko at gcdf.pl (Szymon Olko) Date: Sat, 21 Mar 2009 09:29:38 +0100 Subject: [Freeswitch-users] proper way to clean source tree after build? In-Reply-To: <20090321021028.GA5399@jdc.jasonjgw.net> References: <20090321021028.GA5399@jdc.jasonjgw.net> Message-ID: <49C4A572.9070803@gcdf.pl> Jason White pisze: > I'm building under Debian. > > To clean the source tree after building, I've tried > make clean > debuild clean > > but what often happens is that after updating to a later svn revision and then > trying to build, I get various errors during the compilation process that > disappear if I make a clean checkout from the repository and run the build > again. > > I have also found that using svn export to reproduce the tree in another > directory, then building from that directory, doesn't always solve the problem. > > Is there a better way to clean up the source tree after updating it, to avoid > these issues? I assume there are extraneous modifications or files that simply > aren't being eliminated by the "clean" makefile targets. > > There is 'make current' which make cleaning, uninstalling, updating, building and installing new version. I looked into make file, and I'm doing all without uninstall and install, which I call later when it is possible for me. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Sun Mar 22 23:31:00 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 23 Mar 2009 17:31:00 +1100 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles Message-ID: <20090323063100.GA5058@jdc.jasonjgw.net> I have TLS enabled in my internal and internal-ipv6 profiles as per the stock configuration. When FreeSWITCH is started, sometimes either of the profiles fails to initialize, with an "Unable to create SIP UA for profile" error in the log. If I then start the profile manually sofia profile start from fs_cli, the profile starts up as it should. So far, this has only occurred with the profiles for which I have TLS enabled. I can do more testing to see whether that's part of the problem. Meanwhile, has anyone else seen this? It's revision 12701 from svn. From zhaoxxqq at 163.com Mon Mar 23 02:17:20 2009 From: zhaoxxqq at 163.com (zhaoxxqq) Date: Mon, 23 Mar 2009 17:17:20 +0800 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 Message-ID: <200903231717175620353@163.com> HI, friend, I added to my sip profile in external , like below. --> but. the problem is still exist. Can you help me. Zhao Xiaoqiang ------------------------------------------------------------------------------------------------------------ Hi - It looks like you're getting digits both in the RTP stream and as SIP INFO. Try adding to the SIP profile you're using for inbound calls. --Dave > I'm newbie of FS. I have setup the FS. Most is perfect. I connect FS > to PSTN with DID numbers. For inband I connect it to FS's demo_ivr. > When I press the key, the FS accept always DOUBLE of key. The debug > information like below. > > 2009-03-22 17:50:26 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 > 2009-03-22 17:50:26 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file > 2009-03-22 17:50:26 [DEBUG] switch_ivr_menu.c:308 play_and_collect() waiting for 3/4 digits t/o 2000 > 2009-03-22 17:50:26 [DEBUG] sofia.c:3753 sofia_handle_sip_i_info() INFO DTMF(1) > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:353 play_and_collect() digits '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_menu.c:523 switch_ivr_menu_execute() IVR menu 'jtq_greating' caught invalid input '11' > 2009-03-22 17:50:28 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-03-22 17:50:28 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/external/13323015 > > > Can any friend can help me? > > Zhao Xiaoqiang > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/af706274/attachment-0002.html From Claudio.Cavalera at italtel.it Mon Mar 23 02:17:53 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Mon, 23 Mar 2009 10:17:53 +0100 Subject: [Freeswitch-users] Cepstral and RSS feeds In-Reply-To: <22628039.post@talk.nabble.com> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > Yes that page was perfect, ty. > > (Cepstral 5.1) As far as I know the use of Cepstral 5.x is discouraged and it should be better to stick with Cepstral 4. Unfortunately Cepstral 4 voices are hard to find. :-\ BRs, Claudio Internet E. Mail Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. From brian at freeswitch.org Mon Mar 23 04:53:49 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 06:53:49 -0500 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles In-Reply-To: <20090323063100.GA5058@jdc.jasonjgw.net> References: <20090323063100.GA5058@jdc.jasonjgw.net> Message-ID: You didn't include enough info ... Distro, OS or sofia log... It could be many things... socket lingering is my best guess. /b On Mar 23, 2009, at 1:31 AM, Jason White wrote: > I have TLS enabled in my internal and internal-ipv6 profiles as per > the stock > configuration. > > When FreeSWITCH is started, sometimes either of the profiles fails to > initialize, with an "Unable to create SIP UA for profile" error in > the log. If > I then start the profile manually > sofia profile start > from fs_cli, the profile starts up as it should. > > So far, this has only occurred with the profiles for which I have > TLS enabled. > I can do more testing to see whether that's part of the problem. > > Meanwhile, has anyone else seen this? > > It's revision 12701 from svn. From brian at freeswitch.org Mon Mar 23 04:54:34 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 06:54:34 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: <200903231717175620353@163.com> References: <200903231717175620353@163.com> Message-ID: <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> Tell your phone to stop sending INFO and 2833 at the same time and the problem will stop. /b On Mar 23, 2009, at 4:17 AM, zhaoxxqq wrote: > HI, friend, > I added to my sip profile > in external , like below. > > --> > > > > > > > > but. the problem is still exist. Can you help me. > > Zhao Xiaoqiang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/17036583/attachment-0002.html From dave at 3c.co.uk Mon Mar 23 05:47:16 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 23 Mar 2009 06:47:16 -0600 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> References: <200903231717175620353@163.com> <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> Message-ID: <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> Sorry - my bad - dtmf-type looks like it just controls what's sent, not what's received. Brian's advice is sound, or you can probably work around things right now by editing src/mod/endpoints/mod_sofia/sofia.c - at around line 3838 you'll find: if (dtmf.digit) { /* queue it up */ switch_channel_queue_dtmf(channel, &dtmf); ..more code.. /* Send 200 OK response */ nua_respond(nh, SIP_200_OK, NUTAG_WITH_THIS(nua), TAG_END()); - lose the bit which handles the SIP INFO DTMF by adding a couple of lines thusly: if (dtmf.digit) { #if 0 /* queue it up */ switch_channel_queue_dtmf(channel, &dtmf); ..more code.. #endif /* Send 200 OK response */ nua_respond(nh, SIP_200_OK, NUTAG_WITH_THIS(nua), TAG_END()); It's a nasty hack, but it just might work. --Dave > Tell your phone to stop sending INFO and 2833 at the same time and > the problem will stop. > > /b > > On Mar 23, 2009, at 4:17 AM, zhaoxxqq wrote: > >> HI, friend, >> I added to my sip >> profile in external , like below. >> >> --> >> >> >> >> >> >> >> >> but. the problem is still exist. Can you help me. >> >> Zhao Xiaoqiang > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/851640ad/attachment-0002.html From brian at freeswitch.org Mon Mar 23 06:10:34 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 08:10:34 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> References: <200903231717175620353@163.com> <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> Message-ID: It's more sane to have the phone to NOT send them both in the first place because it is WRONG to send both info and 2833 and NOT totally expect the far end to make heads or tails of it. How about actually have the phone manufacture fix their broken phone? /b On Mar 23, 2009, at 7:47 AM, David Knell wrote: > > It's a nasty hack, but it just might work. > > --Dave From gerry at pstn2.net Mon Mar 23 07:36:41 2009 From: gerry at pstn2.net (Gerry Hull) Date: Mon, 23 Mar 2009 10:36:41 -0400 Subject: [Freeswitch-users] Two portaudio UA's, one FS. Possible? Message-ID: <98a86adf0903230736m66a32ee7u44db2ded80d15c81@mail.gmail.com> I have an app where I would like to run two portaudio user agents on the same computer (two sound cards). I want one UA to be feeding a conference, and the other as a softphone. I don't see a way to run two portaudio ua's on the same instance of FS. Is this possible. If not, OK to run two FS instances? TIA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/a8681a0d/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 23 08:14:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Mar 2009 10:14:44 -0500 Subject: [Freeswitch-users] Two portaudio UA's, one FS. Possible? In-Reply-To: <98a86adf0903230736m66a32ee7u44db2ded80d15c81@mail.gmail.com> References: <98a86adf0903230736m66a32ee7u44db2ded80d15c81@mail.gmail.com> Message-ID: <191c3a030903230814g14533908u5ccc2211a6453303@mail.gmail.com> it's not supported to run 2 instances of PA on 2 soundcards at once in one FS, but it is ok to run 2 difft copies of FS if each one has its own config log and db dirs and none of the configuration collides. 2009/3/23 Gerry Hull > I have an app where I would like to run two portaudio user agents on the > same computer (two sound cards). I want one UA to be feeding a conference, > and the other as a softphone. I don't see a way to run two portaudio ua's > on the same instance of FS. Is this possible. If not, OK to run two FS > instances? > > TIA > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/cc111a45/attachment-0002.html From dave at 3c.co.uk Mon Mar 23 08:33:06 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 23 Mar 2009 09:33:06 -0600 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: References: <200903231717175620353@163.com> <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> Message-ID: <49C7ABB2.80606@3c.co.uk> Hi Brian, > It's more sane to have the phone to NOT send them both in the first > place because it is WRONG to send both info and 2833 and NOT totally > expect the far end to make heads or tails of it. > > How about actually have the phone manufacture fix their broken phone? > In an ideal world, of course; however:- (a) the quick hack is probably a path of lesser resistance to getting Zhao up and running with FS; (b) he said it was an inbound SIP provider, rather than a phone, that he was using, so he'd need to get them to fix their end: might be trivial, might not. --Dave From reply at matthewfong.com Mon Mar 23 08:49:40 2009 From: reply at matthewfong.com (Matthew Fong) Date: Mon, 23 Mar 2009 22:49:40 +0700 Subject: [Freeswitch-users] Another fifo request Message-ID: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/e6462470/attachment-0002.html From mattdfong at gmail.com Mon Mar 23 08:50:16 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Mon, 23 Mar 2009 22:50:16 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> Message-ID: <4256bf830903230850p100b44ffm52e6292fd281aa5@mail.gmail.com> Thanks Anthony, for creating the transfer_after_bridge feature for me. Your rapid actions, are the reason I'm positive I made the right decision switch to to FS. I got another challenge to throw your way. In the current fifo implementation, there's no way to identify which fifo consumer, consumes a caller--besides using other_leg_unique_id on bridge (ie, there's no way to pass data between channels when a fifo bridge is created). I want to be able to transfer some caller information to the consumer channel on bridge, to populate an agent's screen. Under normal (non-fifo) circumstances, when a call is bridged, I've used the 'import' channel variable, so that onBridge, the aleg automatically gets populated with the bleg's 'import' field. However when fifo bridges, it does not recognize import. In other applications, I've gotten around this by using bridge_pre_execute_bleg_app/data to throw a custom event but with fifo, bridge_pre_execute also does not work. I've been looking at the fifo::info event, but again, there's no fifo_action that directly links caller variables and consumer variables together. For now at least, I can get around this by storing uuid information in my separate database, and looking up the other channel's information based on other_leg_unique_id, but it would be nice if I could directly see it when the channel is bridged. Anyway, great program, and I hope you consider implementing these features to make FS even better. Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/44816f63/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 23 08:53:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Mar 2009 10:53:31 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 33, Issue 102 In-Reply-To: <49C7ABB2.80606@3c.co.uk> References: <200903231717175620353@163.com> <4D125312-1693-4B98-A694-EAA9FE5ADF6E@freeswitch.org> <83B2D61C-44B6-4766-A9AE-B30B1030AD8F@3c.co.uk> <49C7ABB2.80606@3c.co.uk> Message-ID: <191c3a030903230853i69bd114dg2fa64ee04ca2976b@mail.gmail.com> c) if we have to, we could add a patch to pick which types of dtmf to accept as well so he could force the equiv of the hack. On Mon, Mar 23, 2009 at 10:33 AM, David Knell wrote: > Hi Brian, > > It's more sane to have the phone to NOT send them both in the first > > place because it is WRONG to send both info and 2833 and NOT totally > > expect the far end to make heads or tails of it. > > > > How about actually have the phone manufacture fix their broken phone? > > > In an ideal world, of course; however:- > (a) the quick hack is probably a path of lesser resistance to getting > Zhao up > and running with FS; > (b) he said it was an inbound SIP provider, rather than a phone, that he > was > using, so he'd need to get them to fix their end: might be trivial, > might not. > > --Dave > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/38d727df/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 23 09:08:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Mar 2009 11:08:05 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> Message-ID: <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> ok, maybe after this i can have a day off ;) 2 variables added to latest trunk: "fifo_caller_consumer_import" "fifo_consumer_caller_import" both work like the regular import but one is a list of vars to copy from caller to consumer and one is a list to copy from consumer to caller. 2009/3/23 Matthew Fong > Thanks Anthony, for creating the transfer_after_bridge feature for me. > Your rapid actions, are the reason I'm positive I made the right decision > switch to to FS. > I got another challenge to throw your way. In the current fifo > implementation, there's no way to identify which fifo consumer, consumes a > caller--besides using other_leg_unique_id on bridge (ie, there's no way to > pass data between channels when a fifo bridge is created). I want to be able > to transfer some caller information to the consumer channel on bridge, to > populate an agent's screen. > > Under normal (non-fifo) circumstances, when a call is bridged, I've used > the 'import' channel variable, so that onBridge, the aleg automatically gets > populated with the bleg's 'import' field. However when fifo bridges, it does > not recognize import. In other applications, I've gotten around this by > using bridge_pre_execute_bleg_app/data to throw a custom event but with > fifo, bridge_pre_execute also does not work. I've been looking at the > fifo::info event, but again, there's no fifo_action that directly links > caller variables and consumer variables together. > > For now at least, I can get around this by storing uuid information in my > separate database, and looking up the other channel's information based > on other_leg_unique_id, but it would be nice if I could directly see it when > the channel is bridged. Anyway, great program, and I hope you consider > implementing these features to make FS even better. Thanks. > > --matt > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/73636456/attachment-0002.html From qulix at mail.ru Mon Mar 23 09:31:59 2009 From: qulix at mail.ru (=?koi8-r?Q?=EC=C5=DB=C1...?=) Date: Mon, 23 Mar 2009 19:31:59 +0300 Subject: [Freeswitch-users] Inbound calls Message-ID: Hi all! There is an error if I try to make inbound call. I don't think it's hard but I'm tired solving it. Any time I call my city number, FS always returns me an error [INCOMPATIBLE_DESTINATION] 2009-03-23 00:29:40 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/mydomain/number-ani at ip [8d0dcff8-1728-11de-b150-e189b327b8ae] 2009-03-23 00:29:40 [NOTICE] sofia.c:2927 sofia_handle_sip_i_state() Hangup sofia/mydomain/number-ani at ip [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-03-23 00:29:40 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 2 (sofia/mydomain/number-ani at ip) Ended 2009-03-23 00:29:40 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/mydomain/number-ani at ip [CS_HANGUP] My extension is: I've made it all the same as http://wiki.freeswitch.org/wiki/Home_PBX_Example (with my own diference line inbound number, dest_number , etc) Could you share some expirience? Give any solution? Or link where I could learn more about it? =\ From brian at freeswitch.org Mon Mar 23 09:39:50 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 11:39:50 -0500 Subject: [Freeswitch-users] Inbound calls In-Reply-To: References: Message-ID: <46DE35FD-1DFC-45F1-9BA6-106EA65022AD@freeswitch.org> make current I think you might have snagged the rev over the weekend while we were fixing a bug. /b On Mar 23, 2009, at 11:31 AM, ????... wrote: > Any time I call my city number, FS always returns me an error > [INCOMPATIBLE_DESTINATION] From codecomplete at free.fr Mon Mar 23 11:45:17 2009 From: codecomplete at free.fr (Gilles) Date: Mon, 23 Mar 2009 19:45:17 +0100 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error Message-ID: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> Hello, This is my first try at compiling Freeswitch (1.0.3). Since I have a Linux box lying around, I'm giving it a shot on a "SUSE Linux Enterprise Desktop 10 SP1 (i586)". Per the instructions in the wiki, I run "make", but get the following error : http://pastebin.ca/1369381 According to the archives, it's likely due to the missing ODBC-devel package. I have a couple of questions: 1. Does Freeswitch require ODBC? 2. Please forgive the newbie question, but... how do I find and install this package on Suse? Thank you. From grevenx at me.com Mon Mar 23 12:01:56 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Mon, 23 Mar 2009 20:01:56 +0100 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error In-Reply-To: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> References: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> Message-ID: 1. FS does not require ODBC, uses SQLite per default How did you run your ./configure ? My guess is that you either have choosen to compile with ODBC, or that the build system somehow detects that you have ODBC installed and tries to compile it with ODBC enabled (and fails). Best regard, Even Andr? Fiskvik On 23. mars. 2009, at 19.45, Gilles wrote: > Hello, > > This is my first try at compiling Freeswitch (1.0.3). Since I have a > Linux box lying around, I'm giving it a shot on a "SUSE Linux > Enterprise Desktop 10 SP1 (i586)". > > Per the instructions in the wiki, I run "make", but get the following > error : http://pastebin.ca/1369381 > > According to the archives, it's likely due to the missing ODBC-devel > package. I have a couple of questions: > 1. Does Freeswitch require ODBC? > 2. Please forgive the newbie question, but... how do I find and > install this package on Suse? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Mar 23 12:22:36 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 14:22:36 -0500 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error In-Reply-To: References: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> Message-ID: <23CC9072-8B9E-4BCB-9B12-91F2FF22BCBE@freeswitch.org> We changed this to autodetect it means you have the libs but no devel headers.. either disable it or install the dev headers. /b On Mar 23, 2009, at 2:01 PM, Even Andr? Fiskvik wrote: > 1. FS does not require ODBC, uses SQLite per default > > How did you run your ./configure ? > My guess is that you either have choosen to compile with ODBC, or that > the > build system somehow detects that you have ODBC installed and tries to > compile it > with ODBC enabled (and fails). > > Best regard, > Even Andr? Fiskvik From raul at etellicom.com Mon Mar 23 12:31:44 2009 From: raul at etellicom.com (Raul Fragoso) Date: Mon, 23 Mar 2009 16:31:44 -0300 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error In-Reply-To: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> References: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> Message-ID: <1237836704.28992.1.camel@raul-laptop> Running the following command as root should install the ODBC development package: yast install unixODBC-devel After that, run configure again and make FS as usual. Regards, Raul On Mon, 2009-03-23 at 19:45 +0100, Gilles wrote: > Hello, > > This is my first try at compiling Freeswitch (1.0.3). Since I have a > Linux box lying around, I'm giving it a shot on a "SUSE Linux > Enterprise Desktop 10 SP1 (i586)". > > Per the instructions in the wiki, I run "make", but get the following > error : http://pastebin.ca/1369381 > > According to the archives, it's likely due to the missing ODBC-devel > package. I have a couple of questions: > 1. Does Freeswitch require ODBC? > 2. Please forgive the newbie question, but... how do I find and > install this package on Suse? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Mar 23 12:43:43 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Mar 2009 15:43:43 -0400 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error In-Reply-To: <1237836704.28992.1.camel@raul-laptop> References: <7.0.1.0.2.20090323193733.02484b40@fredshack.com> <1237836704.28992.1.camel@raul-laptop> Message-ID: <956825C4-39EF-4E12-A96C-0C6B37DB19EE@jerris.com> I still need access to a machine in this state so I can debug and fix this issue. Mike On Mar 23, 2009, at 3:31 PM, Raul Fragoso wrote: > Running the following command as root should install the ODBC > development package: > yast install unixODBC-devel > > After that, run configure again and make FS as usual. > > Regards, > > Raul > > On Mon, 2009-03-23 at 19:45 +0100, Gilles wrote: >> Hello, >> >> This is my first try at compiling Freeswitch (1.0.3). Since I have a >> Linux box lying around, I'm giving it a shot on a "SUSE Linux >> Enterprise Desktop 10 SP1 (i586)". >> >> Per the instructions in the wiki, I run "make", but get the following >> error : http://pastebin.ca/1369381 >> >> According to the archives, it's likely due to the missing ODBC-devel >> package. I have a couple of questions: >> 1. Does Freeswitch require ODBC? >> 2. Please forgive the newbie question, but... how do I find and >> install this package on Suse? >> >> Thank you. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch-users at digitaldan.com Mon Mar 23 14:14:45 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Mon, 23 Mar 2009 15:14:45 -0600 (MDT) Subject: [Freeswitch-users] Sip for Skype Message-ID: <13257395.221237842869198.JavaMail.daniel@radio> You probably already saw this but.... http://www.skypeforsip.com/ Skype is supporting sip for business users. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090323/bdc3369a/attachment-0002.html From pabx_freeswitch at telenet.be Mon Mar 23 15:22:42 2009 From: pabx_freeswitch at telenet.be (henkoegema) Date: Mon, 23 Mar 2009 15:22:42 -0700 (PDT) Subject: [Freeswitch-users] Different files (?) Message-ID: <22670515.post@talk.nabble.com> What is the difference between between the following 3 files: 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 26M 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M Which one should I download, if I want the latest (newest) ? -- View this message in context: http://www.nabble.com/Different-files-%28-%29-tp22670515p22670515.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From pabx_freeswitch at telenet.be Mon Mar 23 15:24:23 2009 From: pabx_freeswitch at telenet.be (henkoegema) Date: Mon, 23 Mar 2009 15:24:23 -0700 (PDT) Subject: [Freeswitch-users] Different files (?) Message-ID: <22670515.post@talk.nabble.com> What is the difference between the following 3 files: 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 26M 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M Which one should I download, if I want the latest (newest) ? -- View this message in context: http://www.nabble.com/Different-files-%28-%29-tp22670515p22670515.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Mar 23 15:34:43 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 17:34:43 -0500 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <22670515.post@talk.nabble.com> References: <22670515.post@talk.nabble.com> Message-ID: <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> I would highly recommend you use the snapshot or svn checkout because we are close to 1.0.4 and it looks like the symlink wasn't updated for 1.0.3 we'll get that fixed. /b On Mar 23, 2009, at 5:24 PM, henkoegema wrote: > 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 > 26M > 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M > 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M From pablosaro at gmail.com Mon Mar 23 16:19:43 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Mon, 23 Mar 2009 20:19:43 -0300 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> References: <22670515.post@talk.nabble.com> <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> Message-ID: <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> BTW, you also recommend SVN trunk for production servers? IMHO, should be a stable release for this purpose. On 3/23/09, Brian West wrote: > I would highly recommend you use the snapshot or svn checkout because > we are close to 1.0.4 and it looks like the symlink wasn't updated for > 1.0.3 we'll get that fixed. > > /b > > On Mar 23, 2009, at 5:24 PM, henkoegema wrote: > >> 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 >> 26M >> 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M >> 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from Gmail for mobile | mobile.google.com From msc at freeswitch.org Mon Mar 23 16:24:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Mar 2009 16:24:49 -0700 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> References: <22670515.post@talk.nabble.com> <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> Message-ID: <87f2f3b90903231624q563b6f1dw1075f30b414f27fe@mail.gmail.com> On Mon, Mar 23, 2009 at 4:19 PM, Pablo Hernan Saro wrote: > BTW, you also recommend SVN trunk for production servers? > IMHO, should be a stable release for this purpose. FreeSWITCH is one of those unusual projects where the SVN trunk is generally more stable than the tagged releases. -MC From brian at freeswitch.org Mon Mar 23 16:26:41 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 18:26:41 -0500 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> References: <22670515.post@talk.nabble.com> <4CD99DCF-5301-4533-AF57-126B78422923@freeswitch.org> <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> Message-ID: <0C8A5F39-868C-478B-931C-B316705EB939@freeswitch.org> Thats up to you... I use SVN Trunk when I know things are good but thats my job... thats why I work hard on QA for the project. We did have some regressions over the past few days that we did finally narrow down and fix over the weekend... and yes we work pretty much 24/7 :) We are trying to get SVN to the point so we can release 1.0.4 and it has MANY fixes over 1.0.3. /b On Mar 23, 2009, at 6:19 PM, Pablo Hernan Saro wrote: > BTW, you also recommend SVN trunk for production servers? > IMHO, should be a stable release for this purpose. > From krice at suspicious.org Mon Mar 23 16:31:07 2009 From: krice at suspicious.org (Ken Rice) Date: Mon, 23 Mar 2009 18:31:07 -0500 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <247f8100903231619g6e5ff3e0n29165fb53cefa2d9@mail.gmail.com> Message-ID: I run SVN on several production servers... "Stable Releases" are just that some point in the code where the maintainers felt it was stable enuff to release on that given day... With FreeSwitch however it never fails, a big bug is found and fixed w/in 2 days of that "stable release" so its not really that stable at all... Until we get more people testing and actually reporting bugs on RCs I doubt this well ever change... > From: Pablo Hernan Saro > Reply-To: > Date: Mon, 23 Mar 2009 20:19:43 -0300 > To: > Subject: Re: [Freeswitch-users] Different files (?) > > BTW, you also recommend SVN trunk for production servers? > IMHO, should be a stable release for this purpose. > > > > On 3/23/09, Brian West wrote: >> I would highly recommend you use the snapshot or svn checkout because >> we are close to 1.0.4 and it looks like the symlink wasn't updated for >> 1.0.3 we'll get that fixed. >> >> /b >> >> On Mar 23, 2009, at 5:24 PM, henkoegema wrote: >> >>> 1. freeswitch-1.0.3.tar.gz 19-Feb-2009 00:34 >>> 26M >>> 2. freeswitch-1.0.latest.tar.gz 24-Jul-2008 12:00 22M >>> 3. freeswitch-snapshot.tar.gz 23-Mar-2009 03:02 49M >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- > Sent from Gmail for mobile | mobile.google.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Mar 23 16:32:58 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Mar 2009 18:32:58 -0500 Subject: [Freeswitch-users] Different files (?) In-Reply-To: References: Message-ID: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> Well put! /b On Mar 23, 2009, at 6:31 PM, Ken Rice wrote: > > Until we get more people testing and actually reporting bugs on RCs > I doubt > this well ever change... From jason at jasonjgw.net Mon Mar 23 16:56:46 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 24 Mar 2009 10:56:46 +1100 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles In-Reply-To: <20090323063100.GA5058@jdc.jasonjgw.net> References: <20090323063100.GA5058@jdc.jasonjgw.net> Message-ID: <20090323235646.GA11567@jdc.jasonjgw.net> Here's a log. OS: Debian Sid, kernel 2.6.26. This happens when FreeSWITCH is started during the boot process from the init scripts, as well as subsequently if it is shut down and restarted. Sometimes it is the internal profile that fails, rather than internal-ipv6, as here. tport_tls_init_master(0x18f6bd0): tls key = /opt/freeswitch/conf/ssl/agent.pem tls_init_context: 140a90a1:SSL routines:(null):(null) tport_listen(0x7fedc0006960): tls_init_master(pf=10 tls/[2001:470:801f::2]:5061): Input/output error nta: bind([2001:470:801f::2]:5061;transport=tls): Input/output error nua: initializing SIP stack failed nua: nua_stack_deinit: entering tport_destroy(0x7fedc0006960) su_epoll_port_deinit(0x18fa810) called 2009-03-24 10:42:12 [ERR] sofia.c:750 sofia_profile_thread_run() Error Creating SIP UA for profile: internal-ipv6 It looks like an operating system issue to me. From mszlazak at aol.com Mon Mar 23 23:40:57 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 24 Mar 2009 02:40:57 -0400 Subject: [Freeswitch-users] setInputCallback not working with Javascript? Message-ID: <8CB7A629F5159D9-17A0-AA5@webmail-de13.sysops.aol.com> I'm getting in build 12653M: [ERR] notify.js:130 mod_spidermonkey()? TypeError: session.setInputCallback is not a function The wiki says this function should work in Javascript. http://wiki.freeswitch.org/wiki/CoreSession_Constructor#session:setInputCallback Also, has there been changes to session.collectInput with type="event"? I get dtmf type events with my callback function but can't seem to get type="event" with speech events. Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/14f2d606/attachment-0002.html From codecomplete at free.fr Tue Mar 24 02:15:09 2009 From: codecomplete at free.fr (Gilles) Date: Tue, 24 Mar 2009 10:15:09 +0100 Subject: [Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error Message-ID: <7.0.1.0.2.20090324095733.0242cf88@fredshack.com> Brian West > We changed this to autodetect it means you have the libs but no devel headers.. either disable it or install the dev headers. Thanks guys, that was it: The stock SLED 10 that I have on this MSI host did have unixODBC.rpm but not its unixODBC-devel.rpm counterpart. I re-read the "Download & Installation Guide" page, which doesn't say that Freeswitch will compile for ODBC instead of SQLite if it detects unixODBC... but doesn't check that its counterpart unixODBC-devel is also installed :-/ I suggest that either the config/make script be updated to check for this, or the wiki be edited to tell users to check for both packages (if N.A. : yast install unixODBC-devel OR rpm -UVH unixODBC-devel.rpm) or configure Freeswitch to use SQLite instead of ODBC (good enough to get newbies started). Cheers, From willbelair at yahoo.com Tue Mar 24 05:51:57 2009 From: willbelair at yahoo.com (Will Smith) Date: Tue, 24 Mar 2009 05:51:57 -0700 (PDT) Subject: [Freeswitch-users] Console Window Message-ID: <996845.60281.qm@web53607.mail.re2.yahoo.com> Hi, I closed the console window that ran FS without shutting down FS. I had to restart the server to get back to the console window. I just wonder if there is a way to bring that window up without restart server. In the file /log/freeswitch.pid, I found a number, is that a seesion id? ? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/a87e6136/attachment-0002.html From dyfet at gnutelephony.org Tue Mar 24 05:54:07 2009 From: dyfet at gnutelephony.org (David Sugar) Date: Tue, 24 Mar 2009 08:54:07 -0400 Subject: [Freeswitch-users] Sip for Skype - g.729 requirement In-Reply-To: <13257395.221237842869198.JavaMail.daniel@radio> References: <13257395.221237842869198.JavaMail.daniel@radio> Message-ID: <49C8D7EF.8050106@gnutelephony.org> They require one use g.729, which is patent encumbered as well as rather computationally intensive. Dan wrote: > You probably already saw this but.... > > http://www.skypeforsip.com/ > > Skype is supporting sip for business users. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: dyfet.vcf Type: text/x-vcard Size: 186 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/6452a43a/attachment-0002.vcf From saeedahmad1981 at gmail.com Tue Mar 24 06:25:08 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Tue, 24 Mar 2009 14:25:08 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <996845.60281.qm@web53607.mail.re2.yahoo.com> References: <996845.60281.qm@web53607.mail.re2.yahoo.com> Message-ID: <19BE75BD4A6543468D8EA83AF14580CC@SaeedLaptop> Did you started FS with -nc option? with this option you can connect to FS using ./fs_cli OR use screen! _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Will Smith Sent: Tuesday, March 24, 2009 1:52 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Console Window Hi, I closed the console window that ran FS without shutting down FS. I had to restart the server to get back to the console window. I just wonder if there is a way to bring that window up without restart server. In the file /log/freeswitch.pid, I found a number, is that a seesion id? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/0ca2c99f/attachment-0002.html From willbelair at yahoo.com Tue Mar 24 06:35:09 2009 From: willbelair at yahoo.com (Will Smith) Date: Tue, 24 Mar 2009 06:35:09 -0700 (PDT) Subject: [Freeswitch-users] Console Window In-Reply-To: <19BE75BD4A6543468D8EA83AF14580CC@SaeedLaptop> Message-ID: <308674.85493.qm@web53611.mail.re2.yahoo.com> No, I started FS with this: /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? ? thank you for help --- On Tue, 3/24/09, Saeed Ahmed wrote: From: Saeed Ahmed Subject: Re: [Freeswitch-users] Console Window To: freeswitch-users at lists.freeswitch.org Date: Tuesday, March 24, 2009, 6:25 AM Did you started FS with ?nc option? with this option you can connect to FS using ./fs_cli OR use screen! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Will Smith Sent: Tuesday, March 24, 2009 1:52 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Console Window ? Hi, I closed the console window that ran FS without shutting down FS. I had to restart the server to get back to the console window. I just wonder if there is a way to bring that window up without restart server. In the file /log/freeswitch.pid, I found a number, is that a seesion id? ? Thank you. ?_______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/1a0bb79f/attachment-0002.html From steve.d.ward at gmail.com Tue Mar 24 06:42:54 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 09:42:54 -0400 Subject: [Freeswitch-users] sip cancel request fails Message-ID: <4ea6e8f20903240642s1c8f46afld96291ca8fb51377@mail.gmail.com> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while the call is still ringing does not work. Why is this request resulting in a 481? I appreciate the help - I'm still just starting to learn SIP & FS. The CANCEL request and 481 response appear as follows on my FS console: recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: ------------------------------------------------------------------------ CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport From: "Steve" >;tag=as7f6965ea To: > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 From: "Steve" >;tag=as7f6965ea To: >;tag=71m745HKHKyjc Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 CSeq: 103 CANCEL Content-Length: 0 -------------------------------------- Thanks. - SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/82662ad8/attachment-0002.html From solko at gcdf.pl Tue Mar 24 06:42:56 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 24 Mar 2009 14:42:56 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <308674.85493.qm@web53611.mail.re2.yahoo.com> References: <308674.85493.qm@web53611.mail.re2.yahoo.com> Message-ID: <49C8E360.8050709@gcdf.pl> Will Smith pisze: > No, I started FS with this: > /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? > run command fs_cli it is almost the same as normal freeswitch console. You just cannot shut it down with '...' command. Just use this one 'fsctl shutdown asap'. > thank you for help > > --- On *Tue, 3/24/09, Saeed Ahmed //* wrote: > > From: Saeed Ahmed > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:25 AM > > Did you started FS with ?nc option? > with this option you can connect to FS using ./fs_cli > OR > use /screen/! > > ------------------------------------------------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Will Smith > *Sent:* Tuesday, March 24, 2009 1:52 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Console Window > > > > Hi, > > I closed the console window that ran FS without shutting down FS. I > had to restart the server to get back to the console window. I just > wonder if there is a way to bring that window up without restart > server. In the file /log/freeswitch.pid, I found a number, is that a > seesion id? > > > > Thank you. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steve.d.ward at gmail.com Tue Mar 24 06:43:12 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 09:43:12 -0400 Subject: [Freeswitch-users] sip cancel request fails Message-ID: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while the call is still ringing does not work. Why is this request resulting in a 481? I appreciate the help - I'm still just starting to learn SIP & FS. The CANCEL request and 481 response appear as follows on my FS console: recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: ------------------------------------------------------------------------ CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport From: "Steve" >;tag=as7f6965ea To: > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 From: "Steve" >;tag=as7f6965ea To: >;tag=71m745HKHKyjc Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 CSeq: 103 CANCEL Content-Length: 0 -------------------------------------- Thanks. - SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/fe31b242/attachment-0002.html From brian at freeswitch.org Tue Mar 24 06:47:01 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Mar 2009 08:47:01 -0500 Subject: [Freeswitch-users] setInputCallback not working with Javascript? In-Reply-To: <8CB7A629F5159D9-17A0-AA5@webmail-de13.sysops.aol.com> References: <8CB7A629F5159D9-17A0-AA5@webmail-de13.sysops.aol.com> Message-ID: <554DE828-B96A-4E17-A974-151CAC50A0E5@freeswitch.org> Javascript doesn't use the Core Session constructor. Its not the same as the other languages. /b On Mar 24, 2009, at 1:40 AM, mszlazak at aol.com wrote: > I'm getting in build 12653M: > > [ERR] notify.js:130 mod_spidermonkey() TypeError: > session.setInputCallback is not a function > > The wiki says this function should work in Javascript. > > http://wiki.freeswitch.org/wiki/ > CoreSession_Constructor#session:setInputCallback > > Also, has there been changes to session.collectInput with > type="event"? I get dtmf type events with my callback function but > can't seem to get type="event" with speech events. > > Mark. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/ad67a971/attachment-0002.html From leon at scarlet-internet.nl Tue Mar 24 06:47:15 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 24 Mar 2009 14:47:15 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <308674.85493.qm@web53611.mail.re2.yahoo.com> References: <308674.85493.qm@web53611.mail.re2.yahoo.com> Message-ID: Just make sure you have mod_event socket loaded in conf/ autoload_modules/modules.conf.xml : And have it configured in conf/autoload_modules/event_socket.conf.xml Then you can use bin/fs_cli to connect to running FS instance. I don't think you can reconnect to a process of which you have disconnected the terminal (without using screen).. regards, Leon On Mar 24, 2009, at 2:35 PM, Will Smith wrote: > No, I started FS with this: > /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? > > thank you for help > > --- On Tue, 3/24/09, Saeed Ahmed wrote: > From: Saeed Ahmed > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:25 AM > > Did you started FS with ?nc option? > with this option you can connect to FS using ./fs_cli > OR > use screen! > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Will Smith > Sent: Tuesday, March 24, 2009 1:52 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Console Window > > > Hi, > > I closed the console window that ran FS without shutting down FS. I > had to restart the server to get back to the console window. I just > wonder if there is a way to bring that window up without restart > server. In the file /log/freeswitch.pid, I found a number, is that a > seesion id? > > > > Thank you. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/eadef7e7/attachment-0002.html From mike at jerris.com Tue Mar 24 06:47:54 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Mar 2009 09:47:54 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> Message-ID: This means we could not match the cancel to a current call dialog. I would need to see the full sip trace of the call to know why, but typically this is because of not matching call Id or to or from tags Mike On Mar 24, 2009, at 9:43 AM, Steven Ward wrote: > A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b- > lab-1) while the call is still ringing does not work. > > Why is this request resulting in a 481? > > I appreciate the help - I'm still just starting to learn SIP & FS. > The CANCEL request and 481 response appear as follows on my FS > console: > > > recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: > > --- > --------------------------------------------------------------------- > CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport > From: "Steve" ;tag=as7f6965ea > To: > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > --- > --------------------------------------------------------------------- > send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: > > --- > --------------------------------------------------------------------- > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 > From: "Steve" ;tag=as7f6965ea > To: ;tag=71m745HKHKyjc > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 > CSeq: 103 CANCEL > Content-Length: 0 > > -------------------------------------- > > > > Thanks. - SW > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/63049bf8/attachment-0002.html From steve.d.ward at gmail.com Tue Mar 24 06:57:10 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 09:57:10 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> Message-ID: <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> Here it is: freeswitch at b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 at 13:53:07.644865: ------------------------------------------------------------------------ OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport From: "Unknown" >;tag=as5adee8f4 To: Contact: > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 24 Mar 2009 13:53:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ------------------------------------------------------------------------ send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060 From: "Unknown" >;tag=as5adee8f4 To: ;tag=DytraHp3K84aD Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 CSeq: 102 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169: ------------------------------------------------------------------------ INVITE sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport From: "Steve" >;tag=as4863e49a To: > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 24 Mar 2009 13:53:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 258 v=0 o=root 4756 4756 IN IP4 10.1.21.44 s=session c=IN IP4 10.1.21.44 t=0 0 m=audio 17956 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 From: "Steve" >;tag=as4863e49a To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Content-Length: 0 ------------------------------------------------------------------------ send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 From: "Steve" >;tag=as4863e49a To: >;tag=e7KHcc76gHUXr Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="10.1.21.44", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103: ------------------------------------------------------------------------ ACK sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport From: "Steve" >;tag=as4863e49a To: >;tag=e7KHcc76gHUXr Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306: ------------------------------------------------------------------------ INVITE sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport From: "Steve" >;tag=as4863e49a To: > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="b-pbx-lab-1", realm="10.1.21.44", algorithm=MD5, uri="sip:70904 at b-pbx-lab-1.mynet.net", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, cnonce="0e89cc90", nc=00000001 Date: Tue, 24 Mar 2009 13:53:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 258 v=0 o=root 4756 4757 IN IP4 10.1.21.44 s=session c=IN IP4 10.1.21.44 t=0 0 m=audio 17956 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 From: "Steve" >;tag=as4863e49a To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Content-Length: 0 ------------------------------------------------------------------------ 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/70904 at 10.1.21.44[1d28557e-187b-11de-8c60-ad87768304bc] 2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing Steve->70904 in context default 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes [1d3a376c-187b-11de-8c60-ad87768304bc] send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291: ------------------------------------------------------------------------ INVITE sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c SIP/2.0 Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p Max-Forwards: 69 From: "Steve" >;tag=gS62F28DB372F To: Call-ID: f4992499-931d-122c-34b1-003018ae1862 CSeq: 112833059 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 328 Remote-Party-ID: "Steve" >;screen=yes;privacy=off v=0 o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 10.1.21.45 s=FreeSWITCH c=IN IP4 10.1.21.45 t=0 0 m=audio 22432 RTP/AVP 0 9 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p Contact: To: ;tag=fa138551 From: "Steve" >;tag=gS62F28DB372F Call-ID: f4992499-931d-122c-34b1-003018ae1862 CSeq: 112833059 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ------------------------------------------------------------------------ 2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:70904 at 10.1.56.106:44952 ;rinstance=481ff1bdc7ab2a4c;fs_nat=yes! send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 From: "Steve" >;tag=as4863e49a To: >;tag=FgDae7QaetHgm Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 sofia_receive_message() Ring-Ready sofia/internal/70904 at 10.1.21.44! 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 switch_ivr_originate() Ring Ready sofia/internal/70904 at 10.1.21.44! recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013: ------------------------------------------------------------------------ CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport From: "Steve" >;tag=as4863e49a To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060 From: "Steve" >;tag=as4863e49a To: >;tag=FgDae7QaetHgm Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 CSeq: 103 CANCEL Content-Length: 0 ------------------------------------------------------------------------ 2009/3/24 Michael Jerris > This means we could not match the cancel to a current call dialog. I > would need to see the full sip trace of the call to know why, but typically > this is because of not matching call Id or to or from tags > > Mike > > > On Mar 24, 2009, at 9:43 AM, Steven Ward wrote: > > A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) > while the call is still ringing does not work. > > Why is this request resulting in a 481? > > I appreciate the help - I'm still just starting to learn SIP & FS. The > CANCEL request and 481 response appear as follows on my FS console: > > > recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: > ------------------------------------------------------------------------ > CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport > From: "Steve" > >;tag=as7f6965ea > To: > > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 > From: "Steve" > >;tag=as7f6965ea > To: > >;tag=71m745HKHKyjc > Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 > CSeq: 103 CANCEL > Content-Length: 0 > -------------------------------------- > > > > Thanks. - SW > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/eca4313e/attachment-0002.html From willbelair at yahoo.com Tue Mar 24 07:23:42 2009 From: willbelair at yahoo.com (Will Smith) Date: Tue, 24 Mar 2009 07:23:42 -0700 (PDT) Subject: [Freeswitch-users] Console Window In-Reply-To: Message-ID: <488035.39134.qm@web53604.mail.re2.yahoo.com> Thank you all for your help. I use : /usr/local/freeswitch/bin/fs_cli? to open a FS instance. But then cannot use "shutdown" command. Then I used 'fsctl shutdown asap'? (directed by Szymon). It works perfectly. ? Again, thank you. Have a great day you all. ? Will --- On Tue, 3/24/09, Leon de Rooij wrote: From: Leon de Rooij Subject: Re: [Freeswitch-users] Console Window To: freeswitch-users at lists.freeswitch.org Date: Tuesday, March 24, 2009, 6:47 AM Just make sure you have mod_event socket loaded in conf/autoload_modules/modules.conf.xml : ?? ? And have it configured in conf/autoload_modules/event_socket.conf.xml Then you can use bin/fs_cli to connect to running FS instance. I don't think you can reconnect to a process of which you have disconnected the terminal (without using screen).. regards, Leon? On Mar 24, 2009, at 2:35 PM, Will Smith wrote: No, I started FS with this: /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? ? thank you for help --- On?Tue, 3/24/09, Saeed Ahmed??wrote: From: Saeed Ahmed Subject: Re: [Freeswitch-users] Console Window To:?freeswitch-users at lists.freeswitch.org Date: Tuesday, March 24, 2009, 6:25 AM Did you started FS with ?nc option? with this option you can connect to FS using ./fs_cli OR? use?screen! From:?freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]?On Behalf Of?Will Smith Sent:?Tuesday, March 24, 2009 1:52 PM To:?freeswitch-users at lists.freeswitch.org Subject:?[Freeswitch-users] Console Window ? Hi, I closed the console window that ran FS without shutting down FS. I had to restart the server to get back to the console window. I just wonder if there is a way to bring that window up without restart server. In the file /log/freeswitch.pid, I found a number, is that a seesion id? ? Thank you. ? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/99f67145/attachment-0002.html From diego.viola at gmail.com Tue Mar 24 07:34:00 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 24 Mar 2009 10:34:00 -0400 Subject: [Freeswitch-users] Console Window In-Reply-To: <488035.39134.qm@web53604.mail.re2.yahoo.com> References: <488035.39134.qm@web53604.mail.re2.yahoo.com> Message-ID: <86a32abc0903240734j46513520k2f7f9b9525c1e18a@mail.gmail.com> I put /usr/local/freeswitch/bin in my path, like this. export PATH=/usr/local/freeswitch/bin:$PATH on my ~/.bashrc, you can put it on your /etc/profile if you want it to be global. Then I just do `freeswitch -nc' when I need to start it, `fs_cli' to connect, and `fsctl shutdown asap' to shut it down, or `freeswitch -stop'. Hope that helps. Diego 2009/3/24 Will Smith : > Thank you all for your help. > I use : /usr/local/freeswitch/bin/fs_cli? to open a FS instance. But then > cannot use "shutdown" command. Then I used 'fsctl shutdown asap'? (directed > by Szymon). It works perfectly. > > Again, thank you. Have a great day you all. > > Will > > --- On Tue, 3/24/09, Leon de Rooij wrote: > > From: Leon de Rooij > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:47 AM > > Just make sure you have mod_event socket loaded in > conf/autoload_modules/modules.conf.xml : > ?? ? > And have it configured in conf/autoload_modules/event_socket.conf.xml > Then you can use bin/fs_cli to connect to running FS instance. > I don't think you can reconnect to a process of which you have disconnected > the terminal (without using screen).. > regards, > Leon > On Mar 24, 2009, at 2:35 PM, Will Smith wrote: > > No, I started FS with this: > /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? > > thank you for help > > --- On?Tue, 3/24/09, Saeed Ahmed??wrote: > > From: Saeed Ahmed > Subject: Re: [Freeswitch-users] Console Window > To:?freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:25 AM > > Did you started FS with ?nc option? > with this option you can connect to FS using ./fs_cli > OR > use?screen! > > ________________________________ > > From:?freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org]?On Behalf Of?Will > Smith > Sent:?Tuesday, March 24, 2009 1:52 PM > To:?freeswitch-users at lists.freeswitch.org > Subject:?[Freeswitch-users] Console Window > > > > Hi, > > I closed the console window that ran FS without shutting down FS. I had to > restart the server to get back to the console window. I just wonder if there > is a way to bring that window up without restart server. In the file > /log/freeswitch.pid, I found a number, is that a seesion id? > > > > Thank you. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From solko at gcdf.pl Tue Mar 24 07:37:07 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 24 Mar 2009 15:37:07 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <488035.39134.qm@web53604.mail.re2.yahoo.com> References: <488035.39134.qm@web53604.mail.re2.yahoo.com> Message-ID: <49C8F013.702@gcdf.pl> Will Smith pisze: > Thank you all for your help. > I use : /usr/local/freeswitch/bin/fs_cli to open a FS instance. But > then cannot use "shutdown" command. Then I used 'fsctl shutdown asap' > (directed by Szymon). It works perfectly. > > Again, thank you. Have a great day you all. > Remeber there are other flags to shutdown freeswitch freeswitch at vertux> fsctl API CALL [fsctl()] output: -USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap|restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration [num]|loglevel [level]] Look for descriptions in wiki. > Will > > --- On *Tue, 3/24/09, Leon de Rooij //* wrote: > > From: Leon de Rooij > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:47 AM > > Just make sure you have mod_event socket loaded in > conf/autoload_modules/modules.conf.xml : > > > > And have it configured in conf/autoload_modules/event_socket.conf.xml > > Then you can use bin/fs_cli to connect to running FS instance. > > I don't think you can reconnect to a process of which you have > disconnected the terminal (without using screen).. > > regards, > > Leon > > On Mar 24, 2009, at 2:35 PM, Will Smith wrote: > >> No, I started FS with this: >> /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? >> >> thank you for help >> >> --- On *Tue, 3/24/09, Saeed Ahmed /> >/* wrote: >> >> From: Saeed Ahmed > > >> Subject: Re: [Freeswitch-users] Console Window >> To: freeswitch-users at lists.freeswitch.org >> >> Date: Tuesday, March 24, 2009, 6:25 AM >> >> Did you started FS with ?nc option? >> with this option you can connect to FS using ./fs_cli >> OR >> use /screen/! >> >> ------------------------------------------------------------------------ >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On >> Behalf Of *Will Smith >> *Sent:* Tuesday, March 24, 2009 1:52 PM >> *To:* freeswitch-users at lists.freeswitch.org >> >> *Subject:* [Freeswitch-users] Console Window >> >> >> >> Hi, >> >> I closed the console window that ran FS without shutting down >> FS. I had to restart the server to get back to the console >> window. I just wonder if there is a way to bring that window >> up without restart server. In the file /log/freeswitch.pid, I >> found a number, is that a seesion id? >> >> >> >> Thank you. >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From grevenx at me.com Tue Mar 24 07:42:44 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Tue, 24 Mar 2009 15:42:44 +0100 Subject: [Freeswitch-users] Console Window In-Reply-To: <488035.39134.qm@web53604.mail.re2.yahoo.com> References: <488035.39134.qm@web53604.mail.re2.yahoo.com> Message-ID: Please see documentation at: http://wiki.freeswitch.org/wiki/Fs_cli and http://wiki.freeswitch.org/wiki/Mod_commands Best regards, Even Andr? On 24. mars. 2009, at 15.23, Will Smith wrote: > Thank you all for your help. > I use : /usr/local/freeswitch/bin/fs_cli to open a FS instance. But > then cannot use "shutdown" command. Then I used 'fsctl shutdown > asap' (directed by Szymon). It works perfectly. > > Again, thank you. Have a great day you all. > > Will > > --- On Tue, 3/24/09, Leon de Rooij wrote: > From: Leon de Rooij > Subject: Re: [Freeswitch-users] Console Window > To: freeswitch-users at lists.freeswitch.org > Date: Tuesday, March 24, 2009, 6:47 AM > > Just make sure you have mod_event socket loaded in conf/ > autoload_modules/modules.conf.xml : > > > > And have it configured in conf/autoload_modules/event_socket.conf.xml > > Then you can use bin/fs_cli to connect to running FS instance. > > I don't think you can reconnect to a process of which you have > disconnected the terminal (without using screen).. > > regards, > > Leon > > On Mar 24, 2009, at 2:35 PM, Will Smith wrote: > >> No, I started FS with this: >> /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? >> >> thank you for help >> >> --- On Tue, 3/24/09, Saeed Ahmed wrote: >> From: Saeed Ahmed >> Subject: Re: [Freeswitch-users] Console Window >> To: freeswitch-users at lists.freeswitch.org >> Date: Tuesday, March 24, 2009, 6:25 AM >> >> Did you started FS with ?nc option? >> with this option you can connect to FS using ./fs_cli >> OR >> use screen! >> >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] On Behalf Of Will Smith >> Sent: Tuesday, March 24, 2009 1:52 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Console Window >> >> >> Hi, >> >> I closed the console window that ran FS without shutting down FS. I >> had to restart the server to get back to the console window. I just >> wonder if there is a way to bring that window up without restart >> server. In the file /log/freeswitch.pid, I found a number, is that >> a seesion id? >> >> >> >> Thank you. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/5339773a/attachment-0002.html From dantavious at comcast.net Tue Mar 24 06:55:47 2009 From: dantavious at comcast.net (Derrick Edwards) Date: Tue, 24 Mar 2009 09:55:47 -0400 Subject: [Freeswitch-users] Sip for Skype - g.729 requirement In-Reply-To: <49C8D7EF.8050106@gnutelephony.org> References: <13257395.221237842869198.JavaMail.daniel@radio> <49C8D7EF.8050106@gnutelephony.org> Message-ID: <1237902947.28021.31.camel@Baby-Girls-Thang> It seems to me that they are just offering SIP trunking like voicepulse. Very interested to see what there rates would be like since they have a large infrastructure. On Tue, 2009-03-24 at 08:54 -0400, David Sugar wrote: > They require one use g.729, which is patent encumbered as well as rather > computationally intensive. > > Dan wrote: > > You probably already saw this but.... > > > > http://www.skypeforsip.com/ > > > > Skype is supporting sip for business users. > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Mar 24 07:54:34 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Mar 2009 10:54:34 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> Message-ID: <32586807-0E8B-4842-976C-97564F67823B@jerris.com> I note that its missing the to tag from the 180 sent 5 seconds earlier (I think thats okay) but the via branch tag is also different, which seems wrong. Can anyone else chime in, I can't recall the dialog matching rules of early dialog like this. Mike On Mar 24, 2009, at 9:57 AM, Steven Ward wrote: > Here it is: > > freeswitch at b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 at > 13:53:07.644865: > > ------------------------------------------------------------------------ > OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport > From: "Unknown" ;tag=as5adee8f4 > To: > Contact: > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Tue, 24 Mar 2009 13:53:07 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060 > From: "Unknown" ;tag=as5adee8f4 > To: ;tag=DytraHp3K84aD > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 > CSeq: 102 OPTIONS > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169: > > ------------------------------------------------------------------------ > INVITE sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport > From: "Steve" ;tag=as4863e49a > To: > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Tue, 24 Mar 2009 13:53:11 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 258 > v=0 > o=root 4756 4756 IN IP4 10.1.21.44 > s=session > c=IN IP4 10.1.21.44 > t=0 0 > m=audio 17956 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > ------------------------------------------------------------------------ > send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 > From: "Steve" ;tag=as4863e49a > To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Content-Length: 0 > > ------------------------------------------------------------------------ > send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660: > > ------------------------------------------------------------------------ > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 > From: "Steve" ;tag=as4863e49a > To: ;tag=e7KHcc76gHUXr > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > Proxy-Authenticate: Digest realm="10.1.21.44", > nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, > qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103: > > ------------------------------------------------------------------------ > ACK sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport > From: "Steve" ;tag=as4863e49a > To: ;tag=e7KHcc76gHUXr > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306: > > ------------------------------------------------------------------------ > INVITE sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport > From: "Steve" ;tag=as4863e49a > To: > Contact: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="b-pbx-lab-1", > realm="10.1.21.44", algorithm=MD5, uri="sip:70904 at b-pbx-lab-1.mynet.net > ", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", > response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, > cnonce="0e89cc90", nc=00000001 > Date: Tue, 24 Mar 2009 13:53:11 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 258 > v=0 > o=root 4756 4757 IN IP4 10.1.21.44 > s=session > c=IN IP4 10.1.21.44 > t=0 0 > m=audio 17956 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > ------------------------------------------------------------------------ > send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526: > > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 > From: "Steve" ;tag=as4863e49a > To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/ > 70904 at 10.1.21.44 [1d28557e-187b-11de-8c60-ad87768304bc] > 2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing Steve->70904 in context default > 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes > [1d3a376c-187b-11de-8c60-ad87768304bc] > send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291: > > ------------------------------------------------------------------------ > INVITE sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c SIP/ > 2.0 > Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p > Max-Forwards: 69 > From: "Steve" ;tag=gS62F28DB372F > To: > Call-ID: f4992499-931d-122c-34b1-003018ae1862 > CSeq: 112833059 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 328 > Remote-Party-ID: "Steve" 70904 at 10.1.21.45>;screen=yes;privacy=off > v=0 > o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 > 10.1.21.45 > s=FreeSWITCH > c=IN IP4 10.1.21.45 > t=0 0 > m=audio 22432 RTP/AVP 0 9 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > ------------------------------------------------------------------------ > recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589: > > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p > Contact: > To: 70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>;tag=fa138551 > From: "Steve";tag=gS62F28DB372F > Call-ID: f4992499-931d-122c-34b1-003018ae1862 > CSeq: 112833059 INVITE > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes > ! > send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060: > > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 > From: "Steve" ;tag=as4863e49a > To: ;tag=FgDae7QaetHgm > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include- > session-description, presence.winfo, message-summary, refer > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 > sofia_receive_message() Ring-Ready sofia/internal/70904 at 10.1.21.44! > 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 > switch_ivr_originate() Ring Ready sofia/internal/70904 at 10.1.21.44! > recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013: > > ------------------------------------------------------------------------ > CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport > From: "Steve" ;tag=as4863e49a > To: > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618: > > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060 > From: "Steve" ;tag=as4863e49a > To: ;tag=FgDae7QaetHgm > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 CANCEL > Content-Length: 0 > > ------------------------------------------------------------------------ > > > > 2009/3/24 Michael Jerris > This means we could not match the cancel to a current call dialog. > I would need to see the full sip trace of the call to know why, but > typically this is because of not matching call Id or to or from tags > > Mike > > > On Mar 24, 2009, at 9:43 AM, Steven Ward > wrote: > >> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b- >> lab-1) while the call is still ringing does not work. >> >> Why is this request resulting in a 481? >> >> I appreciate the help - I'm still just starting to learn SIP & FS. >> The CANCEL request and 481 response appear as follows on my FS >> console: >> >> >> recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: >> >> ------------------------------------------------------------------------ >> CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport >> From: "Steve" ;tag=as7f6965ea >> To: >> Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >> CSeq: 103 CANCEL >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: >> >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 >> From: "Steve" ;tag=as7f6965ea >> To: ;tag=71m745HKHKyjc >> Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >> CSeq: 103 CANCEL >> Content-Length: 0 >> >> -------------------------------------- >> >> >> >> Thanks. - SW >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/fd0f983a/attachment-0002.html From leon at scarlet-internet.nl Tue Mar 24 07:55:37 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 24 Mar 2009 15:55:37 +0100 Subject: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user Message-ID: Hi, I'm trying to get some cli commands working in combination with xml- curl. Endpoints are parsed properly for SIP registrations and invites, but when I use the CLI command "user_exists" it returns false, while I do return an endpoint (same syntax as for a sofia_reg_parse_auth event) on the webserver. Should switch_xml_locate_user event receive a different syntax ? freeswitch at internal> user_exists accountcode gigaset toyos.nl false XML returned from webserver:
thanks & regards, Leon From mike at jerris.com Tue Mar 24 07:59:45 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Mar 2009 10:59:45 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <32586807-0E8B-4842-976C-97564F67823B@jerris.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> <32586807-0E8B-4842-976C-97564F67823B@jerris.com> Message-ID: <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> This appears to be a bug in FreeSWITCH. Can you please test this on current svn trunk and if it is still a problem, please report this as a bug to http://jira.freeswitch.org. MIke On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote: > I note that its missing the to tag from the 180 sent 5 seconds > earlier (I think thats okay) but the via branch tag is also > different, which seems wrong. Can anyone else chime in, I can't > recall the dialog matching rules of early dialog like this. > > Mike > > On Mar 24, 2009, at 9:57 AM, Steven Ward wrote: > >> Here it is: >> >> freeswitch at b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 >> at 13:53:07.644865: >> >> ------------------------------------------------------------------------ >> OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport >> From: "Unknown" ;tag=as5adee8f4 >> To: >> Contact: >> Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 >> CSeq: 102 OPTIONS >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Date: Tue, 24 Mar 2009 13:53:07 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Supported: replaces >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060 >> From: "Unknown" ;tag=as5adee8f4 >> To: ;tag=DytraHp3K84aD >> Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 >> CSeq: 102 OPTIONS >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: 100rel, timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, include- >> session-description, presence.winfo, message-summary, refer >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169: >> >> ------------------------------------------------------------------------ >> INVITE sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport >> From: "Steve" ;tag=as4863e49a >> To: >> Contact: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Date: Tue, 24 Mar 2009 13:53:11 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Supported: replaces >> Content-Type: application/sdp >> Content-Length: 258 >> v=0 >> o=root 4756 4756 IN IP4 10.1.21.44 >> s=session >> c=IN IP4 10.1.21.44 >> t=0 0 >> m=audio 17956 RTP/AVP 0 8 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> ------------------------------------------------------------------------ >> send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 102 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660: >> >> ------------------------------------------------------------------------ >> SIP/2.0 407 Proxy Authentication Required >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: ;tag=e7KHcc76gHUXr >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 102 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: 100rel, timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, include- >> session-description, presence.winfo, message-summary, refer >> Proxy-Authenticate: Digest realm="10.1.21.44", >> nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, >> qop="auth" >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103: >> >> ------------------------------------------------------------------------ >> ACK sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport >> From: "Steve" ;tag=as4863e49a >> To: ;tag=e7KHcc76gHUXr >> Contact: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 102 ACK >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306: >> >> ------------------------------------------------------------------------ >> INVITE sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport >> From: "Steve" ;tag=as4863e49a >> To: >> Contact: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 INVITE >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Proxy-Authorization: Digest username="b-pbx-lab-1", >> realm="10.1.21.44", algorithm=MD5, uri="sip:70904 at b-pbx-lab-1.mynet.net >> ", nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", >> response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, >> cnonce="0e89cc90", nc=00000001 >> Date: Tue, 24 Mar 2009 13:53:11 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Supported: replaces >> Content-Type: application/sdp >> Content-Length: 258 >> v=0 >> o=root 4756 4757 IN IP4 10.1.21.44 >> s=session >> c=IN IP4 10.1.21.44 >> t=0 0 >> m=audio 17956 RTP/AVP 0 8 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> ------------------------------------------------------------------------ >> send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 >> switch_channel_set_name() New Channel sofia/internal/ >> 70904 at 10.1.21.44 [1d28557e-187b-11de-8c60-ad87768304bc] >> 2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() >> Processing Steve->70904 in context default >> 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 >> switch_channel_set_name() New Channel sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes >> [1d3a376c-187b-11de-8c60-ad87768304bc] >> send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291: >> >> ------------------------------------------------------------------------ >> INVITE sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c >> SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p >> Max-Forwards: 69 >> From: "Steve" ;tag=gS62F28DB372F >> To: >> Call-ID: f4992499-931d-122c-34b1-003018ae1862 >> CSeq: 112833059 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: 100rel, timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, include- >> session-description, presence.winfo, message-summary, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 328 >> Remote-Party-ID: "Steve" > 70904 at 10.1.21.45>;screen=yes;privacy=off >> v=0 >> o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 >> 10.1.21.45 >> s=FreeSWITCH >> c=IN IP4 10.1.21.45 >> t=0 0 >> m=audio 22432 RTP/AVP 0 9 8 3 101 13 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589: >> >> ------------------------------------------------------------------------ >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p >> Contact: >> To: > 70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c>;tag=fa138551 >> From: "Steve";tag=gS62F28DB372F >> Call-ID: f4992499-931d-122c-34b1-003018ae1862 >> CSeq: 112833059 INVITE >> User-Agent: X-Lite release 1011s stamp 41150 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-03-24 09:53:11 [NOTICE] sofia.c:2782 >> sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes >> ! >> send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060: >> >> ------------------------------------------------------------------------ >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: ;tag=FgDae7QaetHgm >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: 100rel, timer, precondition, path, replaces >> Allow-Events: talk, presence, dialog, call-info, sla, include- >> session-description, presence.winfo, message-summary, refer >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 >> sofia_receive_message() Ring-Ready sofia/internal/70904 at 10.1.21.44! >> 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 >> switch_ivr_originate() Ring Ready sofia/internal/70904 at 10.1.21.44! >> recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013: >> >> ------------------------------------------------------------------------ >> CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport >> From: "Steve" ;tag=as4863e49a >> To: >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 CANCEL >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618: >> >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060 >> From: "Steve" ;tag=as4863e49a >> To: ;tag=FgDae7QaetHgm >> Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 >> CSeq: 103 CANCEL >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> >> >> >> 2009/3/24 Michael Jerris >> This means we could not match the cancel to a current call dialog. >> I would need to see the full sip trace of the call to know why, but >> typically this is because of not matching call Id or to or from tags >> >> Mike >> >> >> On Mar 24, 2009, at 9:43 AM, Steven Ward >> wrote: >> >>> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b- >>> lab-1) while the call is still ringing does not work. >>> >>> Why is this request resulting in a 481? >>> >>> I appreciate the help - I'm still just starting to learn SIP & >>> FS. The CANCEL request and 481 response appear as follows on my >>> FS console: >>> >>> >>> recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: >>> >>> ------------------------------------------------------------------------ >>> CANCEL sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 >>> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport >>> From: "Steve" ;tag=as7f6965ea >>> To: >>> Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >>> CSeq: 103 CANCEL >>> User-Agent: Asterisk PBX >>> Max-Forwards: 70 >>> Content-Length: 0 >>> >>> ------------------------------------------------------------------------ >>> send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 481 Call/Transaction Does Not Exist >>> Via: SIP/2.0/UDP >>> 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 >>> From: "Steve" ;tag=as7f6965ea >>> To: ;tag=71m745HKHKyjc >>> Call-ID: 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >>> CSeq: 103 CANCEL >>> Content-Length: 0 >>> >>> -------------------------------------- >>> >>> >>> >>> Thanks. - SW >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/ecceb34b/attachment-0002.html From steve.d.ward at gmail.com Tue Mar 24 08:06:15 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 11:06:15 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> <32586807-0E8B-4842-976C-97564F67823B@jerris.com> <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> Message-ID: <4ea6e8f20903240806p45016e6fs90e44512437a8060@mail.gmail.com> Mike, Thanks for taking the time to look at this - I appreciate it. I'll go ahead and test it out on the current svn trunk. - Steve 2009/3/24 Michael Jerris > This appears to be a bug in FreeSWITCH. Can you please test this on > current svn trunk and if it is still a problem, please report this as a bug > to http://jira.freeswitch.org. > MIke > > On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote: > > I note that its missing the to tag from the 180 sent 5 seconds earlier (I > think thats okay) but the via branch tag is also different, which seems > wrong. Can anyone else chime in, I can't recall the dialog matching rules > of early dialog like this. > Mike > > On Mar 24, 2009, at 9:57 AM, Steven Ward wrote: > > Here it is: > > freeswitch at b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 at > 13:53:07.644865: > ------------------------------------------------------------------------ > OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport > From: "Unknown" > >;tag=as5adee8f4 > To: > Contact: > > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Tue, 24 Mar 2009 13:53:07 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > ------------------------------------------------------------------------ > send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060 > From: "Unknown" > >;tag=as5adee8f4 > To: ;tag=DytraHp3K84aD > Call-ID: 2e6222b16df27200056f742a070f0b56 at 10.1.21.44 > CSeq: 102 OPTIONS > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Length: 0 > ------------------------------------------------------------------------ > recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169: > ------------------------------------------------------------------------ > INVITE sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport > From: "Steve" > >;tag=as4863e49a > To: > > > Contact: > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Tue, 24 Mar 2009 13:53:11 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 258 > v=0 > o=root 4756 4756 IN IP4 10.1.21.44 > s=session > c=IN IP4 10.1.21.44 > t=0 0 > m=audio 17956 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Content-Length: 0 > ------------------------------------------------------------------------ > send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660: > ------------------------------------------------------------------------ > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > >;tag=e7KHcc76gHUXr > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Proxy-Authenticate: Digest realm="10.1.21.44", > nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth" > Content-Length: 0 > ------------------------------------------------------------------------ > recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103: > ------------------------------------------------------------------------ > ACK sip:70904 at b-pbx-lab-1.mynet.net SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport > From: "Steve" > >;tag=as4863e49a > To: > >;tag=e7KHcc76gHUXr > Contact: > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > ------------------------------------------------------------------------ > recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306: > ------------------------------------------------------------------------ > INVITE sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport > From: "Steve" > >;tag=as4863e49a > To: > > > Contact: > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="b-pbx-lab-1", realm="10.1.21.44", > algorithm=MD5, uri="sip:70904 at b-pbx-lab-1.mynet.net", > nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", > response="f632ad9dd89f761cbfa442d7ed9c5556", qop=auth, cnonce="0e89cc90", > nc=00000001 > Date: Tue, 24 Mar 2009 13:53:11 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 258 > v=0 > o=root 4756 4757 IN IP4 10.1.21.44 > s=session > c=IN IP4 10.1.21.44 > t=0 0 > m=audio 17956 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.686526: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() > New Channel sofia/internal/70904 at 10.1.21.44[1d28557e-187b-11de-8c60-ad87768304bc] > 2009-03-24 09:53:11 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() > Processing Steve->70904 in context default > 2009-03-24 09:53:11 [NOTICE] switch_channel.c:567 switch_channel_set_name() > New Channel sofia/internal/ > sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes[1d3a376c-187b-11de-8c60-ad87768304bc] > send 1212 bytes to udp/[10.1.56.106]:44952 at 13:53:11.814291: > ------------------------------------------------------------------------ > INVITE sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c SIP/2.0 > Via: SIP/2.0/UDP 10.1.21.45;rport;branch=z9hG4bKDyS5SjU3vK33p > Max-Forwards: 69 > From: "Steve" > >;tag=gS62F28DB372F > To: > Call-ID: f4992499-931d-122c-34b1-003018ae1862 > CSeq: 112833059 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 328 > Remote-Party-ID: "Steve" > >;screen=yes;privacy=off > v=0 > o=FreeSWITCH 5141707032885022242 491120215176734726 IN IP4 10.1.21.45 > s=FreeSWITCH > c=IN IP4 10.1.21.45 > t=0 0 > m=audio 22432 RTP/AVP 0 9 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > ------------------------------------------------------------------------ > recv 424 bytes from udp/[10.1.56.106]:44952 at 13:53:11.916589: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.1.21.45;rport=5060;branch=z9hG4bKDyS5SjU3vK33p > Contact: > To: >;tag=fa138551 > From: "Steve" > >;tag=gS62F28DB372F > Call-ID: f4992499-931d-122c-34b1-003018ae1862 > CSeq: 112833059 INVITE > User-Agent: X-Lite release 1011s stamp 41150 > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-03-24 09:53:11 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/ > sip:70904 at 10.1.56.106:44952;rinstance=481ff1bdc7ab2a4c;fs_nat=yes! > send 729 bytes to udp/[10.1.21.44]:5060 at 13:53:12.011060: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > >;tag=FgDae7QaetHgm > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: 100rel, timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-03-24 09:53:12 [NOTICE] mod_sofia.c:1287 sofia_receive_message() > Ring-Ready sofia/internal/70904 at 10.1.21.44! > 2009-03-24 09:53:12 [NOTICE] switch_ivr_originate.c:1692 > switch_ivr_originate() Ring Ready sofia/internal/70904 at 10.1.21.44! > recv 362 bytes from udp/[10.1.21.44]:5060 at 13:53:17.063013: > ------------------------------------------------------------------------ > CANCEL sip:70904 at b-pbx-lab-1.mynet.netSIP/2.0 > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport > From: "Steve" > >;tag=as4863e49a > To: > > > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > ------------------------------------------------------------------------ > send 327 bytes to udp/[10.1.21.44]:5060 at 13:53:17.063618: > ------------------------------------------------------------------------ > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK00d6d874;rport=5060 > From: "Steve" > >;tag=as4863e49a > To: > >;tag=FgDae7QaetHgm > Call-ID: 4db5b31f3f9d99c436804e4b54277b3e at 10.1.21.44 > CSeq: 103 CANCEL > Content-Length: 0 > ------------------------------------------------------------------------ > > > > 2009/3/24 Michael Jerris > >> This means we could not match the cancel to a current call dialog. I >> would need to see the full sip trace of the call to know why, but typically >> this is because of not matching call Id or to or from tags >> >> Mike >> >> >> On Mar 24, 2009, at 9:43 AM, Steven Ward wrote: >> >> A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) >> while the call is still ringing does not work. >> >> Why is this request resulting in a 481? >> >> I appreciate the help - I'm still just starting to learn SIP & FS. The >> CANCEL request and 481 response appear as follows on my FS console: >> >> >> recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616: >> >> ------------------------------------------------------------------------ >> CANCEL sip: >> 70904 at b-pbx-lab-1.mynet.net SIP/2.0 >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport >> From: "Steve" 70904 at 10.1.21.44 >> >;tag=as7f6965ea >> To: 70904 at b-lab-1.mynet.net> >> Call-ID: <237598fd102b739a03b4a4047bf69843 at 10.1.21.44> >> 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >> CSeq: 103 CANCEL >> User-Agent: Asterisk PBX >> Max-Forwards: 70 >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235: >> >> ------------------------------------------------------------------------ >> SIP/2.0 481 Call/Transaction Does Not Exist >> Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060 >> From: "Steve" 70904 at 10.1.21.44 >> >;tag=as7f6965ea >> To: 70904 at b-lab-1.mynet.net >> >;tag=71m745HKHKyjc >> Call-ID: <237598fd102b739a03b4a4047bf69843 at 10.1.21.44> >> 237598fd102b739a03b4a4047bf69843 at 10.1.21.44 >> CSeq: 103 CANCEL >> Content-Length: 0 >> -------------------------------------- >> >> >> >> Thanks. - SW >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/5ecb5fd8/attachment-0002.html From woof at nortel.com Tue Mar 24 09:32:16 2009 From: woof at nortel.com (Andy Spitzer) Date: Tue, 24 Mar 2009 12:32:16 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: <4ea6e8f20903240806p45016e6fs90e44512437a8060@mail.gmail.com> References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> <32586807-0E8B-4842-976C-97564F67823B@jerris.com> <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> <4ea6e8f20903240806p45016e6fs90e44512437a8060@mail.gmail.com> Message-ID: Woof! Appears to be a recently fixed * bug: 0014431: Bad branch parameter value in CANCEL request http://bugs.digium.com/view.php?id=14431 --Woof! From dujinfang at gmail.com Tue Mar 24 10:30:57 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 25 Mar 2009 01:30:57 +0800 Subject: [Freeswitch-users] FreeSWITCH Chinese Community Message-ID: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> Hi all, In about a year of playing with FreeSWITCH, I really like it. Not only the software is great but also the community. As people keep asking me where can find some documentation in Chinese, I told them if they want to play in deep they need to know English better. However, the fact is, even one can read English without problem, he still prefer find some information in his native language first. So, I'd like to start a Chinese community hoping FreeSWITCH can get more popular in China. As we saw big telecom companies merged in the last year, I bet VoIP technology and application will go more and more faster in the near future... While FS official site has plenty of documentation, obviously we don't want to translate word by word. Then where should we start from? A forum? There was a big argue between a forum and mailing list in the last few weeks, and finally an English forum and an Italian one is out. While I can find server in China, the major pain is that running any kind of BBS in China mainland need to get some kind of permissions by the government first. Any idea, suggestion? Anyone want to help or cooperate about this? And, if I can get this runing, can I use the FreeSWITCH logo and www.freeswitch.org.cn ? Best regards, Seven. From msc at freeswitch.org Tue Mar 24 11:03:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Mar 2009 11:03:51 -0700 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> Message-ID: <87f2f3b90903241103l30302179u655713047b0d4bf9@mail.gmail.com> > While FS official site has plenty of documentation, obviously we don't > want to translate word by word. ?Then where should we start from? A > forum? I would start by finding as many people as possible who are literate in both English and Chinese and who are willing to help out. Once you have your group assembled you will have a better idea of what your goals should be. More people helping will make things easier to accomplish. There was a big argue between a forum and mailing list in the > last few weeks, and finally an English forum and an Italian one is > out. ?While I can find server in China, the major pain is that running > any kind of BBS in China mainland need to get some kind of permissions > by the government first. > > Any idea, suggestion? Anyone want to help or cooperate about this? I am looking into creating a multi-language wiki at wiki.freeswitch.org. Anyone with experience in setting up multiple languages with MediaWiki software please contact me. So far we have people willing to help create documentation in French, Spanish, Portugese, Russian, and now Chinese. I think we have some Italians out there as well! (ciao bella) Please email me off list if you are in a position to assist with the wiki and languages other than English. > > And, if I can get this runing, can I use the FreeSWITCH logo and www.freeswitch.org.cn > ?? Please direct questions about the FreeSWITCH logo and domain names to consulting at freeswitch.org. FreeSWITCH and the logo are trademarks so it would be best to get permission from Anthony before doing anything. -MC From mitul at enterux.com Tue Mar 24 11:32:20 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 25 Mar 2009 00:02:20 +0530 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> Message-ID: Hello, I can provide you with the hosting on our box in US let me know. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 24-Mar-09, at 23:00, dujinfang wrote: > Hi all, > > In about a year of playing with FreeSWITCH, I really like it. Not only > the software is great but also the community. As people keep asking me > where can find some documentation in Chinese, I told them if they want > to play in deep they need to know English better. However, the fact > is, even one can read English without problem, he still prefer find > some information in his native language first. So, I'd like to start a > Chinese community hoping FreeSWITCH can get more popular in China. As > we saw big telecom companies merged in the last year, I bet VoIP > technology and application will go more and more faster in the near > future... > > While FS official site has plenty of documentation, obviously we don't > want to translate word by word. Then where should we start from? A > forum? There was a big argue between a forum and mailing list in the > last few weeks, and finally an English forum and an Italian one is > out. While I can find server in China, the major pain is that running > any kind of BBS in China mainland need to get some kind of permissions > by the government first. > > Any idea, suggestion? Anyone want to help or cooperate about this? > > And, if I can get this runing, can I use the FreeSWITCH logo and www.freeswitch.org.cn > ? > > Best regards, > Seven. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Mar 24 11:42:33 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Mar 2009 13:42:33 -0500 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> Message-ID: <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> How about we all work together and work on the FreeSWITCH.org infrastructure instead of spreading the resources thinner and thinner till nobody is doing really much of anything. We need people to step up and help out with the website, wiki, jira, irc, testing and various other things that currently are spread thin. ;) Please!!! I'm not asking everyone to step up and code in C... /b On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote: > Hello, > > I can provide you with the hosting on our box in US let me know. > > Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt Ltd, > The Enterprise Linux Company(r), > http://www.enterux.com/ > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/d846c84c/attachment-0002.html From steve.d.ward at gmail.com Tue Mar 24 11:45:01 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Tue, 24 Mar 2009 14:45:01 -0400 Subject: [Freeswitch-users] sip cancel request fails In-Reply-To: References: <4ea6e8f20903240643w46459c6cu337ffb61bd146594@mail.gmail.com> <4ea6e8f20903240657o5b97afdag342bab018afb51e8@mail.gmail.com> <32586807-0E8B-4842-976C-97564F67823B@jerris.com> <3E8F659A-2A0D-48FF-BCC5-DB470E1513B1@jerris.com> <4ea6e8f20903240806p45016e6fs90e44512437a8060@mail.gmail.com> Message-ID: <4ea6e8f20903241145t70aeb0asd37376429367974a@mail.gmail.com> Ah ha! Thanks for finding that. I updated my * server and I'm all set. Many thanks for all the feedback and help. 2009/3/24 Andy Spitzer > Woof! > > Appears to be a recently fixed * bug: > > 0014431: Bad branch parameter value in CANCEL request > http://bugs.digium.com/view.php?id=14431 > > --Woof! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090324/742e974c/attachment-0002.html From mitul at enterux.com Tue Mar 24 11:56:09 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 25 Mar 2009 00:26:09 +0530 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> Message-ID: <9DBFC7A7-DA73-4CF2-AC91-4BF3EF914884@enterux.com> Brian, I can help with website, wiki and testing, tell me what's next step forward. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 25-Mar-09, at 0:12, Brian West wrote: > How about we all work together and work on the FreeSWITCH.org > infrastructure instead of spreading the resources thinner and > thinner till nobody is doing really much of anything. > > We need people to step up and help out with the website, wiki, jira, > irc, testing and various other things that currently are spread > thin. ;) Please!!! I'm not asking everyone to step up and code in > C... > > /b > > On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote: > >> Hello, >> >> I can provide you with the hosting on our box in US let me know. >> >> Regards, >> Mitul Limbani, >> Founder & CEO, >> Enterux Solutions Pvt Ltd, >> The Enterprise Linux Company(r), >> http://www.enterux.com/ >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/b6d78784/attachment-0002.html From msc at freeswitch.org Tue Mar 24 12:05:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Mar 2009 12:05:21 -0700 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <9DBFC7A7-DA73-4CF2-AC91-4BF3EF914884@enterux.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> <9DBFC7A7-DA73-4CF2-AC91-4BF3EF914884@enterux.com> Message-ID: <87f2f3b90903241205g72570c3w92c99317d2c86fc5@mail.gmail.com> 2009/3/24 Mitul Limbani : > Brian, > I can help with website, wiki and testing, tell me what's next step forward. > Mitul, do you have any experience with MediaWiki? -MC From dujinfang at gmail.com Tue Mar 24 17:42:44 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 25 Mar 2009 08:42:44 +0800 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <87f2f3b90903241103l30302179u655713047b0d4bf9@mail.gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <87f2f3b90903241103l30302179u655713047b0d4bf9@mail.gmail.com> Message-ID: <0C9F4BE9-F6AE-444B-993C-FF093DBA9CE0@gmail.com> On Mar 25, 2009, at 2:03 AM, Michael Collins wrote: >> While FS official site has plenty of documentation, obviously we >> don't >> want to translate word by word. Then where should we start from? A >> forum? > > I would start by finding as many people as possible who are literate > in both English and Chinese and who are willing to help out. Once you > have your group assembled you will have a better idea of what your > goals should be. More people helping will make things easier to > accomplish. Ppl not only literate in both English and Chinse, but also need to has good knowledge of VoIP and FreeSWITCH. > There was a big argue between a forum and mailing list in the >> last few weeks, and finally an English forum and an Italian one is >> out. While I can find server in China, the major pain is that >> running >> any kind of BBS in China mainland need to get some kind of >> permissions >> by the government first. >> >> Any idea, suggestion? Anyone want to help or cooperate about this? > > I am looking into creating a multi-language wiki at > wiki.freeswitch.org. Anyone with experience in setting up multiple > languages with MediaWiki software please contact me. So far we have > people willing to help create documentation in French, Spanish, > Portugese, Russian, and now Chinese. I think we have some Italians out > there as well! (ciao bella) > > Please email me off list if you are in a position to assist with the > wiki and languages other than English. > How's the plan of a multi-language wiki? I think mediaWiKi do support multi-language. So, maybe we can start from wiki.freeswitch.org/zh_CN/ or some structure else. While it is possible to follow the change log of the EN wiki to change Chinese version accordingly, it would be challenge to make sure the consistency. And regardlessly what if Chinese version change first? I'd like to help with this. But I cannot translate all hundreds of pages and make it consistency by my self. But I'm pretty sure that we would get a group of ppl to do this. So the start maybe either translate some key pages and make the consistency, or translate as much pages ... Email me off list on how to start this. :) >> >> And, if I can get this runing, can I use the FreeSWITCH logo and www.freeswitch.org.cn >> ? > > Please direct questions about the FreeSWITCH logo and domain names to > consulting at freeswitch.org. FreeSWITCH and the logo are trademarks so > it would be best to get permission from Anthony before doing anything. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Tue Mar 24 17:47:02 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 25 Mar 2009 08:47:02 +0800 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> Message-ID: <63686540-6117-4B42-AF21-25A5659DC959@gmail.com> On Mar 25, 2009, at 2:42 AM, Brian West wrote: > How about we all work together and work on the FreeSWITCH.org > infrastructure instead of spreading the resources thinner and > thinner till nobody is doing really much of anything. > I agree with this and would like to work with the wiki.freeswitch.org on the EN and CN language part. However, I don't think we can maintain a Chinese version of IRC, seems still we need a Chinese BBS. > We need people to step up and help out with the website, wiki, jira, > irc, testing and various other things that currently are spread > thin. ;) Please!!! I'm not asking everyone to step up and code in > C... > Take me in :) > /b > > On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote: > >> Hello, >> >> I can provide you with the hosting on our box in US let me know. >> >> Regards, >> Mitul Limbani, >> Founder & CEO, >> Enterux Solutions Pvt Ltd, >> The Enterprise Linux Company(r), >> http://www.enterux.com/ >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/d78c79c3/attachment-0002.html From brian at freeswitch.org Tue Mar 24 17:53:53 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Mar 2009 19:53:53 -0500 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <63686540-6117-4B42-AF21-25A5659DC959@gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> <63686540-6117-4B42-AF21-25A5659DC959@gmail.com> Message-ID: <4F0F01A9-39DC-4234-8301-6996F9BC9255@freeswitch.org> On Mar 24, 2009, at 7:47 PM, dujinfang wrote: > > On Mar 25, 2009, at 2:42 AM, Brian West wrote: >> How about we all work together and work on the FreeSWITCH.org >> infrastructure instead of spreading the resources thinner and >> thinner till nobody is doing really much of anything. >> > > I agree with this and would like to work with the > wiki.freeswitch.org on the EN and CN language part. However, I don't > think we can maintain a Chinese version of IRC, seems still we need > a Chinese BBS. Someone tried to do a spanish IRC channel it doesn't have anyone in it but me most of the time... hrm...... I'm not trying to discount anything here but it seems language specific anything seems to not take off. We need more cheerleaders.... btw cluecon is coming up.... remember to register early! ;) >> We need people to step up and help out with the website, wiki, >> jira, irc, testing and various other things that currently are >> spread thin. ;) Please!!! I'm not asking everyone to step up and >> code in C... >> > > Take me in :) What do you want to be involved in? #freeswitch-web is the correct channel to be in if you want to help out on IRC. ;) > >> /b From msc at freeswitch.org Tue Mar 24 18:01:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Mar 2009 18:01:44 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: ClueCon 2009 - Register Today! Message-ID: <87f2f3b90903241801q71b33a80n1cbf5cbd525e86ae@mail.gmail.com> The FreeSWITCH Team is pleased to announce that all may register for ClueCon 2009 immediately! ClueCon is the Telephony Developers Conference by developers, for developers. This year's event will be held at the beautiful Wyndham Hotel in Chicago, August 4-6. Special room rates have been secured and you will enjoy staying at this luxurious hotel. The conference attendee price is only $499 per person, but hurry! This early bird special price is valid only until June 30th. The schedule is still being finalized, however you can expect to hear from notable names in the world of telephony. You will also have opportunities to spend one-on-one time with developers, vendors, and other attendees. One special highlight for this year is the great MacBook Pro giveaway! On Thursday at the end of the conference a 15" MacBook Pro will be given away to one lucky attendee. This laptop will be laser engraved with the logos of all of the sponsors for ClueCon 2009 and will be used as the presenters' laptop for the entire conference. Start making your travel plans right away. Visit http://www.cluecon.com and sign up today. We look forward to seeing you in Chicago this August! Michael S Collins 877-742-CLUE From mitul at enterux.com Tue Mar 24 19:30:25 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 25 Mar 2009 08:00:25 +0530 Subject: [Freeswitch-users] FreeSWITCH Chinese Community In-Reply-To: <87f2f3b90903241205g72570c3w92c99317d2c86fc5@mail.gmail.com> References: <946658EE-82F7-48AA-8D5E-96FF8397C1E7@gmail.com> <6BB08FE6-BE0E-4CC1-8AD3-7463457BDEF3@freeswitch.org> <9DBFC7A7-DA73-4CF2-AC91-4BF3EF914884@enterux.com> <87f2f3b90903241205g72570c3w92c99317d2c86fc5@mail.gmail.com> Message-ID: <74493056-0544-4833-BF9A-90CFBBCC90B4@enterux.com> MC, I don't have much exp with mediawiki but have some with drupal. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 25-Mar-09, at 0:35, Michael Collins wrote: > 2009/3/24 Mitul Limbani : >> Brian, >> I can help with website, wiki and testing, tell me what's next step >> forward. >> > > Mitul, do you have any experience with MediaWiki? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raul at etellicom.com Tue Mar 24 22:53:30 2009 From: raul at etellicom.com (Raul Fragoso) Date: Wed, 25 Mar 2009 02:53:30 -0300 Subject: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user In-Reply-To: References: Message-ID: <1237960410.18715.0.camel@raul-laptop> You should use the id value of the user, not accountcode: user_exists id gigaset toyos.nl Regards, Raul On Tue, 2009-03-24 at 15:55 +0100, Leon de Rooij wrote: > Hi, > > I'm trying to get some cli commands working in combination with xml- > curl. > > Endpoints are parsed properly for SIP registrations and invites, but > when I use the CLI command "user_exists" it returns false, while I do > return an endpoint (same syntax as for a sofia_reg_parse_auth event) > on the webserver. > > Should switch_xml_locate_user event receive a different syntax ? > > freeswitch at internal> user_exists accountcode gigaset toyos.nl > false > > XML returned from webserver: > > > >
> > > > value="a211336e29c4f3756e6af343ce6da27b"/> > > > > > > > > > > > > > > > > > >
> > > thanks & regards, > > Leon > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mrene_lists at avgs.ca Tue Mar 24 22:57:07 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 25 Mar 2009 01:57:07 -0400 Subject: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user In-Reply-To: <1237960410.18715.0.camel@raul-laptop> References: <1237960410.18715.0.camel@raul-laptop> Message-ID: As Raul said, the user_exists function will look for a user *attribute* so unless you write your user as , you need to use id (and the later wouldnt set the accountcode variable, no) Math On 25-Mar-09, at 1:53 AM, Raul Fragoso wrote: > You should use the id value of the user, not accountcode: > > user_exists id gigaset toyos.nl > > Regards, > > Raul > > On Tue, 2009-03-24 at 15:55 +0100, Leon de Rooij wrote: >> Hi, >> >> I'm trying to get some cli commands working in combination with xml- >> curl. >> >> Endpoints are parsed properly for SIP registrations and invites, but >> when I use the CLI command "user_exists" it returns false, while I do >> return an endpoint (same syntax as for a sofia_reg_parse_auth event) >> on the webserver. >> >> Should switch_xml_locate_user event receive a different syntax ? >> >> freeswitch at internal> user_exists accountcode gigaset toyos.nl >> false >> >> XML returned from webserver: >> >> >> >>
>> >> >> >> > value="a211336e29c4f3756e6af343ce6da27b"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> >> >> thanks & regards, >> >> Leon >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From codecomplete at free.fr Wed Mar 25 02:51:32 2009 From: codecomplete at free.fr (Gilles) Date: Wed, 25 Mar 2009 10:51:32 +0100 Subject: [Freeswitch-users] [Remote SIP client] Couple of questions Message-ID: <7.0.1.0.2.20090325104634.02701c88@fredshack.com> Hello, I have a couple of questions related to having SIP users connecting from the Net to a Freeswitch server through NAT routers on both ends: 1. How must I configure routers on both ends? I understand that I need to route incoming TCP/UDP 5080 into the Freeswitch server, but what about the other router? I guess I also need to route this port to let the SIP phone ring, but what about data (RTP/RTCP)? 2. The Freeswitch server is connected to the POTS with either an OpenVox PCI card or a Linksys 3102 box: When a call is made between the POTS and a remote SIP phone (ie. out there on the Net, not on the same LAN as the Freeswitch server), is there a way for data to flow directly from the POTS to the remote SIP client instead of through the Freeswitch server? Thank you. From leon at scarlet-internet.nl Wed Mar 25 03:15:01 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Wed, 25 Mar 2009 11:15:01 +0100 Subject: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user In-Reply-To: References: <1237960410.18715.0.camel@raul-laptop> Message-ID: <9B6C044E-B6EC-48A9-B711-0E3A3AD48142@scarlet-internet.nl> Thanks.. I see it's even documented at http://wiki.freeswitch.org/wiki/Mod_commands#user_exists (doh) regards, Leon On Mar 25, 2009, at 6:57 AM, Mathieu Rene wrote: > As Raul said, the user_exists function will look for a user > *attribute* so unless you write your user as accountcode="blah2">, you need to use id (and the later wouldnt set > the accountcode variable, no) > > Math > > On 25-Mar-09, at 1:53 AM, Raul Fragoso wrote: > >> You should use the id value of the user, not accountcode: >> >> user_exists id gigaset toyos.nl >> >> Regards, >> >> Raul >> >> On Tue, 2009-03-24 at 15:55 +0100, Leon de Rooij wrote: >>> Hi, >>> >>> I'm trying to get some cli commands working in combination with xml- >>> curl. >>> >>> Endpoints are parsed properly for SIP registrations and invites, but >>> when I use the CLI command "user_exists" it returns false, while I >>> do >>> return an endpoint (same syntax as for a sofia_reg_parse_auth >>> event) >>> on the webserver. >>> >>> Should switch_xml_locate_user event receive a different syntax ? >>> >>> freeswitch at internal> user_exists accountcode gigaset toyos.nl >>> false >>> >>> XML returned from webserver: >>> >>> >>> >>>
>>> >>> >>> >>> >> value="a211336e29c4f3756e6af343ce6da27b"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>>
>>> >>> >>> thanks & regards, >>> >>> Leon >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From leon at scarlet-internet.nl Wed Mar 25 03:16:54 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Wed, 25 Mar 2009 11:16:54 +0100 Subject: [Freeswitch-users] user_data sends two header_strings named key Message-ID: Hi, I think this is a bug in mod_commands.c : When using "user_data" function from mod_commands (mod_commands.c, line 358) A header string "key" is added to params (mod_commands.c, line 362) Then switch_xml_locate_user is called (switch_xml.c, line 1712) And that function adds another header string "key" (switch_xml.c, line 1740) This results in two keys with the same name being sent ( to in my case mod_xml_curl ). I opened an issue in jira for this (is it alright like that ?) : http://jira.freeswitch.org/browse/MODAPP-242 regards, Leon From codecomplete at free.fr Wed Mar 25 04:08:34 2009 From: codecomplete at free.fr (Gilles) Date: Wed, 25 Mar 2009 12:08:34 +0100 Subject: [Freeswitch-users] Starting Freeswitch at boot-time with rc.d script? Message-ID: <7.0.1.0.2.20090325120411.024773f0@fredshack.com> Hello I'm struggling to have Freeswitch start automatically in a Suse host at boot-time through the rc.d script provided in SVN. Here's what I did so far: 1. cp /usr/src/freeswitch/build/freeswitch.init.suse /etc/init.d/freeswitch 2. chmod 755 /etc/init.d/freeswitch 3. chkconfig freeswitch 345 4. chkconfig -l freeswitch 5. ln -s /usr/sbin/rcfreeswitch /etc/init.d/freeswitch 6. Edited /etc/init.d/freeswitch to correct the following two lines: #FREESWITCH_BIN=/opt/freeswitch/bin/freeswitch FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitch #FREESWITCH_CONFIG=/etc/sysconfig/freeswitch FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml 7. Ran the script: /etc/init.d/freeswitch start [...] Session Rate[30] SQL [Enabled] 2009-03-25 11:54:56 [WARNING] switch_console.c:432 console_thread() We've become an orphan, no more console for us. => According to the FAQ, running Freeswitch without a console requires using the "-nc" switch. Should I edit the rc.d script further to use this, or is the issue elsewhere? Thank you. From codecomplete at free.fr Wed Mar 25 04:56:55 2009 From: codecomplete at free.fr (Gilles) Date: Wed, 25 Mar 2009 12:56:55 +0100 Subject: [Freeswitch-users] Starting Freeswitch at boot-time with rc.d script? Message-ID: <7.0.1.0.2.20090325125455.02842df0@free.fr> Add-on: The script is almost fine now, but I don't know whether the script really requires a config file, and if yes, where it can be found (the XML file doesn't seem to be it): FREESWITCH_PARAMS="-nc" FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitch #BAD!!! FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml #. $FREESWITCH_CONFIG Thank you. From pablosaro at gmail.com Wed Mar 25 05:20:19 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Wed, 25 Mar 2009 09:20:19 -0300 Subject: [Freeswitch-users] Console Window In-Reply-To: References: <488035.39134.qm@web53604.mail.re2.yahoo.com> Message-ID: <247f8100903250520v1fb568bbh99d47c4bd88e0e40@mail.gmail.com> Also guys, I recommend to use a script to handle FS instead of directly executing the binary. FS provides some examples. On 3/24/09, Even Andr? Fiskvik wrote: > Please see documentation at: > http://wiki.freeswitch.org/wiki/Fs_cli > and > http://wiki.freeswitch.org/wiki/Mod_commands > > > Best regards, > Even Andr? > > On 24. mars. 2009, at 15.23, Will Smith wrote: > >> Thank you all for your help. >> I use : /usr/local/freeswitch/bin/fs_cli to open a FS instance. But >> then cannot use "shutdown" command. Then I used 'fsctl shutdown >> asap' (directed by Szymon). It works perfectly. >> >> Again, thank you. Have a great day you all. >> >> Will >> >> --- On Tue, 3/24/09, Leon de Rooij wrote: >> From: Leon de Rooij >> Subject: Re: [Freeswitch-users] Console Window >> To: freeswitch-users at lists.freeswitch.org >> Date: Tuesday, March 24, 2009, 6:47 AM >> >> Just make sure you have mod_event socket loaded in conf/ >> autoload_modules/modules.conf.xml : >> >> >> >> And have it configured in conf/autoload_modules/event_socket.conf.xml >> >> Then you can use bin/fs_cli to connect to running FS instance. >> >> I don't think you can reconnect to a process of which you have >> disconnected the terminal (without using screen).. >> >> regards, >> >> Leon >> >> On Mar 24, 2009, at 2:35 PM, Will Smith wrote: >> >>> No, I started FS with this: >>> /usr/local/freeswitch/bin/freeswitch . What can I do in this case ? >>> >>> thank you for help >>> >>> --- On Tue, 3/24/09, Saeed Ahmed wrote: >>> From: Saeed Ahmed >>> Subject: Re: [Freeswitch-users] Console Window >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Tuesday, March 24, 2009, 6:25 AM >>> >>> Did you started FS with ?nc option? >>> with this option you can connect to FS using ./fs_cli >>> OR >>> use screen! >>> >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>> ] On Behalf Of Will Smith >>> Sent: Tuesday, March 24, 2009 1:52 PM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: [Freeswitch-users] Console Window >>> >>> >>> Hi, >>> >>> I closed the console window that ran FS without shutting down FS. I >>> had to restart the server to get back to the console window. I just >>> wonder if there is a way to bring that window up without restart >>> server. In the file /log/freeswitch.pid, I found a number, is that >>> a seesion id? >>> >>> >>> >>> Thank you. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- Sent from Gmail for mobile | mobile.google.com From steve.d.ward at gmail.com Wed Mar 25 05:44:41 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Wed, 25 Mar 2009 08:44:41 -0400 Subject: [Freeswitch-users] conference calling ideas Message-ID: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> If any can share some ideas, I'm looking at making conference calls simple for the end-users of my FS system. Here are some issues I'm kicking around: 1. To create/join conferences - do you make a pre-defined list of extensions, each of which would join the caller to a particular conference room, or do you perhaps define one extension that will join the caller to a dynamically-named conference room (perhaps after making a choice in an IVR)? 2. How do you implement the idea of a conference leader - having a user who must call into a conference first before anybody else can join? Thanks for any ideas. - SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/1c716cfa/attachment-0002.html From solko at gcdf.pl Wed Mar 25 06:02:17 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 25 Mar 2009 14:02:17 +0100 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> Message-ID: <49CA2B59.6070708@gcdf.pl> Steven Ward pisze: > If any can share some ideas, I'm looking at making conference > calls simple for the end-users of my FS system. > > Here are some issues I'm kicking around: > > 1. To create/join conferences - do you make a pre-defined list of > extensions, each of which would join the caller to a particular > conference room, or do you perhaps define one extension that will join > the caller to a dynamically-named conference room (perhaps after making > a choice in an IVR)? > > 2. How do you implement the idea of a conference leader - having a > user who must call into a conference first before anybody else can join? > > Thanks for any ideas. > Can you give more info what you plan to do? If you don't care for conferences name and they can be numbers then just make one extension and then use that extension or part of it to name conference. If you want to make conference leader then make 2 separate extensions, in second extension which is used for not leaders use that conference command. 'conference xxx bgdial '.In current SVN, bgdial does not create new conference it must exists to make that command. This can be make just by dial plan. If you plan to control FS from any language then you have much more abilities. > - SW > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steve.d.ward at gmail.com Wed Mar 25 06:37:45 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Wed, 25 Mar 2009 09:37:45 -0400 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <49CA2B59.6070708@gcdf.pl> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> <49CA2B59.6070708@gcdf.pl> Message-ID: <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> Szymon, I want to provide a service wherein a user can reserve a teleconference room for a partiuclar time and control who can join the conference call (only those invited). I want to support several of such conference calls at any given time. I want callers who were not invited to a conference call, but who try to call in, to be notified they can't join. Also, it should be simple enough that the user just has to dial a number and be able to know what to do from there to join the conference. If the conference leader is late to the conference call and others try to call in (before the conference is created), the callers should be notified the conf leader hasn't started the conference yet. Perhaps they can be told to try later, or they can be put on hold. These are the kinds of things I need to think about implementing, and I do not know yet what other features would be useful. Thanks. On Wed, Mar 25, 2009 at 9:02 AM, Szymon Olko wrote: > Steven Ward pisze: > > If any can share some ideas, I'm looking at making conference > > calls simple for the end-users of my FS system. > > > > Here are some issues I'm kicking around: > > > > 1. To create/join conferences - do you make a pre-defined list of > > extensions, each of which would join the caller to a particular > > conference room, or do you perhaps define one extension that will join > > the caller to a dynamically-named conference room (perhaps after making > > a choice in an IVR)? > > > > 2. How do you implement the idea of a conference leader - having a > > user who must call into a conference first before anybody else can join? > > > > Thanks for any ideas. > > > Can you give more info what you plan to do? > > If you don't care for conferences name and they can be numbers then just > make one extension and then use that extension or part of > it to name conference. > > If you want to make conference leader then make 2 separate extensions, in > second extension which is used for not leaders use that > conference command. 'conference xxx bgdial '.In current SVN, > bgdial does not create new conference it must exists to > make that command. > > This can be make just by dial plan. If you plan to control FS from any > language then you have much more abilities. > > > - SW > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/f779404d/attachment-0002.html From solko at gcdf.pl Wed Mar 25 07:02:32 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 25 Mar 2009 15:02:32 +0100 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> <49CA2B59.6070708@gcdf.pl> <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> Message-ID: <49CA3978.7020704@gcdf.pl> Steven Ward pisze: > Szymon, I want to provide a service wherein a user can reserve a > teleconference room for a partiuclar time and control who can join the > conference call (only those invited). I want to support several of > such conference calls at any given time. > > I want callers who were not invited to a conference call, but who try to > call in, to be notified they can't join. > Reserving conference means some storage of it. I think you have write some application to control it. You can use store that in dialplan but it will be hard to make such dialplan. There is pin code for conference so you can use it. > Also, it should be simple enough that the user just has to dial a number > and be able to know what to do from there to join the conference. > > If the conference leader is late to the conference call and others try > to call in (before the conference is created), the callers should be > notified the conf leader hasn't started the conference yet. Perhaps > they can be told to try later, or they can be put on hold. > > These are the kinds of things I need to think about implementing, and I > do not know yet what other features would be useful. Thanks. I would make it like that. If you need to limit total number of conferences then you have to manage time of reservation. If you don't need that then make one extension in which you do the following. - get pincode from user. - get admin_pincode from user. - look in db for next available conference number - store conference number and pincode in database. - return that number to user. Make second extension where you do folowing: - get conference number from user - get pincode from user - check in db if conference and pincode are the same as stored in db. - check in database if admin is in conference. Make third extension: - get conference number from user - get admin pincode from user - check for matching in db - transfer to conference - store in db that admin is online With this way users must get conference name and pincode to access. You can change it and in first extension add ability for user to add others members with their personal number or something like that. Szymon Olko > On Wed, Mar 25, 2009 at 9:02 AM, Szymon Olko wrote: > > Steven Ward pisze: > > If any can share some ideas, I'm looking at making conference > > calls simple for the end-users of my FS system. > > > > Here are some issues I'm kicking around: > > > > 1. To create/join conferences - do you make a pre-defined list of > > extensions, each of which would join the caller to a particular > > conference room, or do you perhaps define one extension that will join > > the caller to a dynamically-named conference room (perhaps after > making > > a choice in an IVR)? > > > > 2. How do you implement the idea of a conference leader - having a > > user who must call into a conference first before anybody else can > join? > > > > Thanks for any ideas. > > > Can you give more info what you plan to do? > > If you don't care for conferences name and they can be numbers then > just make one extension and then use that extension or part of > it to name conference. > > If you want to make conference leader then make 2 separate > extensions, in second extension which is used for not leaders use that > conference command. 'conference xxx bgdial '.In current > SVN, bgdial does not create new conference it must exists to > make that command. > > This can be make just by dial plan. If you plan to control FS from > any language then you have much more abilities. > > > - SW > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From congxin.zhao at gmail.com Wed Mar 25 03:52:50 2009 From: congxin.zhao at gmail.com (congxin zhao) Date: Wed, 25 Mar 2009 18:52:50 +0800 Subject: [Freeswitch-users] help about the rtp port. In-Reply-To: References: Message-ID: Hi, I meet a issue about the rtp port. The media port value in the sdp of SIP 200 OK which freeswitch internal sends to ua is always 0, so the media forward is always wrong. What could cause this problem possibly? No matter I set or The problem always showes. Does the following rtp-ip and ext-rtp-ip set correct? In external.xml, I set both rtp-ip and ext-rtp-ip as $${external_rtp_ip}, is that correct? I am not clear about this two variables in the external.xml and internal.xml external.xml: external.xml: internal.xml: internal.xml: Thanks, -Congxin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/fbbea456/attachment-0002.html From dujinfang at gmail.com Wed Mar 25 08:25:12 2009 From: dujinfang at gmail.com (dujinfang) Date: Wed, 25 Mar 2009 23:25:12 +0800 Subject: [Freeswitch-users] Starting Freeswitch at boot-time with rc.d script? In-Reply-To: <7.0.1.0.2.20090325125455.02842df0@free.fr> References: <7.0.1.0.2.20090325125455.02842df0@free.fr> Message-ID: <6854FBCC-B8D9-4947-8662-9E1A30680B57@gmail.com> I don't think you need FREESWITCH_CONFIG. It will find all configuration files on default place, say /usr/local/freeswitch/conf If you store config files in other place, the command line should like this /usr/local/freeswitch/bin/freeswitch -conf /tmp/conf -db /tmp/db -log / tmp/log On Mar 25, 2009, at 7:56 PM, Gilles wrote: > Add-on: The script is almost fine now, but I don't know whether the > script really requires a config file, and if yes, where it can be > found (the XML file doesn't seem to be it): > > FREESWITCH_PARAMS="-nc" > FREESWITCH_BIN=/usr/local/freeswitch/bin/freeswitchI > > #BAD!!! FREESWITCH_CONFIG=/usr/local/freeswitch/conf/freeswitch.xml > #. $FREESWITCH_CONFIG > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mattdfong at gmail.com Wed Mar 25 08:26:30 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 25 Mar 2009 22:26:30 +0700 Subject: [Freeswitch-users] Cron-like execution in FS Message-ID: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com> I'm wondering if there's any features that allow the cron-like execution of code inside of Freeswitch, preferably with lua--or if I am stuck using the api interface and running the cron outside of freeswitch. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/45a46f45/attachment-0002.html From pablosaro at gmail.com Wed Mar 25 08:43:38 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Wed, 25 Mar 2009 12:43:38 -0300 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> References: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> Message-ID: <247f8100903250843l9fa97a8oa96adae8c05d6f01@mail.gmail.com> First of all, thanks for your answers. You guys are awesome and FS rocks. Don't take it the wrong way... My custom is to install production systems using stable versions and apply the corresponding security patches as soon as they are released. But I don't change software versions until I really need to do it (for example when I need a new functionality). I've installed production systems using FS SVN trunk, but it changes every day. So, to keep it up to date I have to re-compile it every week at least. I was wondering if there is a version you can affirm has no major bugs. If not, there is no problem with that... Just wondering... Pablo On Mon, Mar 23, 2009 at 8:32 PM, Brian West wrote: > Well put! > > /b > > On Mar 23, 2009, at 6:31 PM, Ken Rice wrote: > >> >> Until we get more people testing and actually reporting bugs on RCs >> I doubt >> this well ever change... > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Mark.Tabron at rnid-typetalk.org.uk Wed Mar 25 08:49:59 2009 From: Mark.Tabron at rnid-typetalk.org.uk (Mark Tabron) Date: Wed, 25 Mar 2009 15:49:59 -0000 Subject: [Freeswitch-users] Problem dialing out via E1 References: <11C1F78E88546B4387E9CC0603051CFE76536C@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765381@tt-mail.RNID.TYPETALK.LOCAL><11C1F78E88546B4387E9CC0603051CFE765382@tt-mail.RNID.TYPETALK.LOCAL> <200903201929.05829.stkn@freeswitch.org> Message-ID: <11C1F78E88546B4387E9CC0603051CFE765389@tt-mail.RNID.TYPETALK.LOCAL> Thanks for the additional info. Found a problem with ozmod_libri.so missing from the mod directory after installation (fixed by manually copying file over) and I can now happily confirm I can make calls both in and out! Thanks for all the help on this - really appreciated. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stefan Knoblich Sent: 20 March 2009 18:29 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Am Friday 20 March 2009 schrieb Mark Tabron: > Installed libpri but I'm stuck on what entries to put in > openzap.conf.xml, here's how I have the span setup at the moment: > > > > > > > > > > > Node and Switch type aren't documented for libpri from what I can tell - > I know the former is either CPE or NET, though, I'm unsure what other > values can be used for switch type. > The value for switch is invalid, it's going to fall back to dms100 with that one set. Valid settings are: ni1, ni2, dms100, euroisdn, lucent5e, att4ess, gr303eoc and gr303tmc and "euroisdn" is the one you'll want for a E1 line. Another setting you may need is: stkn -- ------------------------------------------------------------------------ ------- Stefan Knoblich | axsentis GmbH | Web: http://www.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | http://oss.axsentis.de/ Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------ ------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- From brian at freeswitch.org Wed Mar 25 09:35:54 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 11:35:54 -0500 Subject: [Freeswitch-users] help about the rtp port. In-Reply-To: References: Message-ID: <58A60265-2CBF-409A-8483-7371F82AD4B6@freeswitch.org> Zero for the port means something please provide a complete sip trace. /b On Mar 25, 2009, at 5:52 AM, congxin zhao wrote: > Hi, > > I meet a issue about the rtp port. The media port value in the sdp > of SIP 200 OK which freeswitch internal sends to ua is always 0, so > the media forward is always wrong. What could cause this problem > possibly? > > No matter I set > > or > > The problem always showes. > > Does the following rtp-ip and ext-rtp-ip set correct? In > external.xml, I set both rtp-ip and ext-rtp-ip as $$ > {external_rtp_ip}, is that correct? I am not clear about this two > variables in the external.xml and internal.xml > > external.xml: > external.xml: > > internal.xml: > internal.xml: > > > Thanks, > -Congxin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/ff5f22a3/attachment-0002.html From msc at freeswitch.org Wed Mar 25 09:40:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 09:40:02 -0700 Subject: [Freeswitch-users] Cron-like execution in FS In-Reply-To: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com> References: <4256bf830903250826ta96905dn4c9fee129351208f@mail.gmail.com> Message-ID: <87f2f3b90903250940s4bf14381m8bb78d69f2f922fe@mail.gmail.com> 2009/3/25 Matthew Fong : > I'm wondering if there's any features that allow the cron-like execution of > code inside of Freeswitch,?preferably?with lua--or if I am stuck using the > api interface and running the cron outside of freeswitch. > --matt I guess the most important question you can answer is this: why do you need the cron-like feature to be inside FreeSWITCH? -MC From msc at freeswitch.org Wed Mar 25 10:03:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 10:03:49 -0700 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <247f8100903250843l9fa97a8oa96adae8c05d6f01@mail.gmail.com> References: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> <247f8100903250843l9fa97a8oa96adae8c05d6f01@mail.gmail.com> Message-ID: <87f2f3b90903251003i109fc3c7k3ad129308d6343de@mail.gmail.com> On Wed, Mar 25, 2009 at 8:43 AM, Pablo Hernan Saro wrote: > First of all, thanks for your answers. You guys are awesome and FS > rocks. Don't take it the wrong way... No offense taken. > My custom is to install production systems using stable versions and > apply the corresponding security patches as soon as they are released. > But I don't change software versions until I really need to do it (for > example when I need a new functionality). Nothing wrong with this custom. > I've installed production systems using FS SVN trunk, but it changes > every day. So, to keep it up to date I have to re-compile it every > week at least. Understood. Production systems have their own issues. One custom that is common is to have a production system and a "sandbox" system. You could have your sandbox system to a "make current" every night and you could run simple tests via cron to see if anything important to you is somehow broken. If all is well then you know that you can upgrade your production system. > I was wondering if there is a version you can affirm has no major > bugs. If not, there is no problem with that... Just wondering... This is a tough one. It may be a minor bug to me but it could be a major bug to you. In other words this is almost an impossible question to answer. This much I can say: the FS devs make lots of changes each day but they don't leave the SVN trunk in an unstable state at any point, especially at the end of the day. (GMT -6 time zone.) The bottom line is still the same: FreeSWITCH is one of those rare software projects where the SVN trunk is almost always more stable and less buggy than the most recent "stable" release. My recommendation to you is to find the update schedule that works best for you, be it daily, weekly or some other period. -MC From chris at fowler.cc Wed Mar 25 11:19:09 2009 From: chris at fowler.cc (Chris Fowler) Date: Wed, 25 Mar 2009 11:19:09 -0700 Subject: [Freeswitch-users] DTMF Missing Digits Message-ID: <1238005149.12393.1307283305@webmail.messagingengine.com> Any thoughts on why FS saw all digits "1029" but only reports '029'? 2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 play_and_collect() digits '029' Config: Trace: 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 1:2000 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1301 do_2833() Send start packet for [1] ts=1129880426 dur=160/160/2000 seq=2804 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=320/320/2000 seq=2805 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=480/480/2000 seq=2806 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=640/640/2000 seq=2807 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=800/800/2000 seq=2808 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=960/960/2000 seq=2809 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1120/1120/2000 seq=2810 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1280/1280/2000 seq=2811 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1440/1440/2000 seq=2812 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1600/1600/2000 seq=2813 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1760/1760/2000 seq=2814 2009-03-25 10:48:41 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [1] ts=1129880426 dur=1920/1920/2000 seq=2815 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [1] ts=1129880426 dur=2080/2080/2000 seq=2816 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [1] ts=1129880426 dur=2080/2080/2000 seq=2817 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [1] ts=1129880426 dur=2080/2080/2000 seq=2818 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 0:2160 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1301 do_2833() Send start packet for [0] ts=1129884426 dur=160/160/2160 seq=2819 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=320/320/2160 seq=2820 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=480/480/2160 seq=2821 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=640/640/2160 seq=2822 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=800/800/2160 seq=2823 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=960/960/2160 seq=2824 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1120/1120/2160 seq=2825 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1280/1280/2160 seq=2826 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1440/1440/2160 seq=2827 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1600/1600/2160 seq=2828 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1760/1760/2160 seq=2829 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=1920/1920/2160 seq=2830 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [0] ts=1129884426 dur=2080/2080/2160 seq=2831 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [0] ts=1129884426 dur=2240/2240/2160 seq=2832 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [0] ts=1129884426 dur=2240/2240/2160 seq=2833 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [0] ts=1129884426 dur=2240/2240/2160 seq=2834 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 2:2000 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1301 do_2833() Send start packet for [2] ts=1129887626 dur=160/160/2000 seq=2835 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=320/320/2000 seq=2836 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=480/480/2000 seq=2837 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=640/640/2000 seq=2838 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=800/800/2000 seq=2839 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=960/960/2000 seq=2840 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1120/1120/2000 seq=2841 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1280/1280/2000 seq=2842 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1440/1440/2000 seq=2843 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1600/1600/2000 seq=2844 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1760/1760/2000 seq=2845 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 0:2080 2009-03-25 10:48:42 [DEBUG] switch_ivr_play_say.c:1280 switch_ivr_play_file() done playing file 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [2] ts=1129887626 dur=1920/1920/2000 seq=2846 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [2] ts=1129887626 dur=2080/2080/2000 seq=2847 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [2] ts=1129887626 dur=2080/2080/2000 seq=2848 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [2] ts=1129887626 dur=2080/2080/2000 seq=2849 2009-03-25 10:48:42 [DEBUG] switch_ivr_menu.c:319 play_and_collect() waiting for 3/4 digits t/o 1500 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 9:2000 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1301 do_2833() Send start packet for [9] ts=1129890826 dur=160/160/2000 seq=2850 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=320/320/2000 seq=2851 2009-03-25 10:48:42 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=480/480/2000 seq=2852 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=640/640/2000 seq=2853 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=800/800/2000 seq=2854 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=960/960/2000 seq=2855 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1120/1120/2000 seq=2856 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1280/1280/2000 seq=2857 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1440/1440/2000 seq=2858 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1600/1600/2000 seq=2859 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1760/1760/2000 seq=2860 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send middle packet for [9] ts=1129890826 dur=1920/1920/2000 seq=2861 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [9] ts=1129890826 dur=2080/2080/2000 seq=2862 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [9] ts=1129890826 dur=2080/2080/2000 seq=2863 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1240 do_2833() Send end packet for [9] ts=1129890826 dur=2080/2080/2000 seq=2864 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 2:2080 2009-03-25 10:48:43 [DEBUG] switch_rtp.c:1786 switch_rtp_dequeue_dtmf() RTP RECV DTMF 9:2080 2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 play_and_collect() digits '029' From brian at freeswitch.org Wed Mar 25 11:27:18 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 13:27:18 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238005149.12393.1307283305@webmail.messagingengine.com> References: <1238005149.12393.1307283305@webmail.messagingengine.com> Message-ID: <7BE6FEF3-E132-4DA2-939B-85D6590DA757@freeswitch.org> First off what SVN rev? Remember when reporting issues try to include all the information you can! /b On Mar 25, 2009, at 1:19 PM, Chris Fowler wrote: > Any thoughts on why FS saw all digits "1029" but only reports '029'? > 2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 > play_and_collect() > digits '029' From chris at fowler.cc Wed Mar 25 11:49:45 2009 From: chris at fowler.cc (Chris Fowler) Date: Wed, 25 Mar 2009 11:49:45 -0700 Subject: [Freeswitch-users] DTMF Missing Digits Message-ID: <1238006985.18397.1307291893@webmail.messagingengine.com> >> First off what SVN rev? Remember when reporting issues try to include all the information you can! Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647) From brian at freeswitch.org Wed Mar 25 11:53:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 13:53:21 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238006985.18397.1307291893@webmail.messagingengine.com> References: <1238006985.18397.1307291893@webmail.messagingengine.com> Message-ID: <4604EEB6-4F7D-427A-9C19-15DC3BF3CB8B@freeswitch.org> Please review this link http://wiki.freeswitch.org/wiki/Reporting_Bugs The rules are try to reproduce this on SVN Trunk... I am pretty sure we fixed this one already. /b On Mar 25, 2009, at 1:49 PM, Chris Fowler wrote: > Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647) From pablosaro at gmail.com Wed Mar 25 11:55:06 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Wed, 25 Mar 2009 15:55:06 -0300 Subject: [Freeswitch-users] Different files (?) In-Reply-To: <87f2f3b90903251003i109fc3c7k3ad129308d6343de@mail.gmail.com> References: <361508B1-55EC-4680-98D5-9F15E76230CC@freeswitch.org> <247f8100903250843l9fa97a8oa96adae8c05d6f01@mail.gmail.com> <87f2f3b90903251003i109fc3c7k3ad129308d6343de@mail.gmail.com> Message-ID: <247f8100903251155k41472db1ja353ffe6676cd5e7@mail.gmail.com> Hi Michael, All I have to say is thank you very much. Thanks for this explanation, that's what I was looking for. So, I will figure out which is the most convenient update schedule for me and I will build a sandbox for this exclusive purpose. Not only FS rocks, but the community too. Kudos Pablo On Wed, Mar 25, 2009 at 2:03 PM, Michael Collins wrote: > On Wed, Mar 25, 2009 at 8:43 AM, Pablo Hernan Saro wrote: >> First of all, thanks for your answers. You guys are awesome and FS >> rocks. Don't take it the wrong way... > > No offense taken. > >> My custom is to install production systems using stable versions and >> apply the corresponding security patches as soon as they are released. >> But I don't change software versions until I really need to do it (for >> example when I need a new functionality). > > Nothing wrong with this custom. > >> I've installed production systems using FS SVN trunk, but it changes >> every day. So, to keep it up to date I have to re-compile it every >> week at least. > > Understood. Production systems have their own issues. One custom that > is common is to have a production system and a "sandbox" system. You > could have your sandbox system to a "make current" every night and you > could run simple tests via cron to see if anything important to you is > somehow broken. If all is well then you know that you can upgrade your > production system. > >> I was wondering if there is a version you can affirm has no major >> bugs. If not, there is no problem with that... Just wondering... > > This is a tough one. It may be a minor bug to me but it could be a > major bug to you. In other words this is almost an impossible question > to answer. This much I can say: the FS devs make lots of changes each > day but they don't leave the SVN trunk in an unstable state at any > point, especially at the end of the day. (GMT -6 time zone.) > > The bottom line is still the same: FreeSWITCH is one of those rare > software projects where the SVN trunk is almost always more stable and > less buggy than the most recent "stable" release. My recommendation to > you is to find the update schedule that works best for you, be it > daily, weekly or some other period. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jforman at wcgltd.com Wed Mar 25 12:01:01 2009 From: jforman at wcgltd.com (Josh Forman) Date: Wed, 25 Mar 2009 15:01:01 -0400 Subject: [Freeswitch-users] Question about FAQ question Message-ID: I'm running Freeswitch on Ubuntu 64bit Intrepid and on a svn rev 12722, freeswitch would install and run fine but as soon as calls started coming in it would have a segmentation fault. This is the second svn snapshot I've had this happen on. In the Freeswitch FAQs there is a question concerning segmentation fault on ubuntu 64bit except they say it occurs on start. Their solution is to recompile libedit which I plan to try regardless, but I just wanted to know if the scenario they refer to is the same as what I'm experiencing or was there some other problem where freeswitch would segfault while initially loading. Until then I'm still running rev 12289 on the system that is having issues. Thanks -Josh From brian at freeswitch.org Wed Mar 25 12:07:24 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 14:07:24 -0500 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: References: Message-ID: On Mar 25, 2009, at 2:01 PM, Josh Forman wrote: > I'm running Freeswitch on Ubuntu 64bit Intrepid and on a svn rev > 12722, freeswitch would install and run fine but as soon as calls > started coming in it would have a segmentation fault. This is the > second svn snapshot I've had this happen on. Please review http://wiki.freeswitch.org/wiki/Reporting_Bugs Collect the info and report it on jira. If it still happens on SVN trunk as of NOW then please collect the back trace as per the reporting bugs guide and we'll investigate the issue. Also be aware if you aren't doing a "make current", you could have build skew which could be the whole problem all along. Incorrectly updating your system will result in all kinds of strange behaviors. > In the Freeswitch FAQs there is a question concerning segmentation > fault on ubuntu 64bit except they say it occurs on start. Their > solution is to recompile libedit which I plan to try regardless, but I > just wanted to know if the scenario they refer to is the same as what > I'm experiencing or was there some other problem where freeswitch > would segfault while initially loading. I don't think this is the problem you're having. > > Until then I'm still running rev 12289 on the system that is having > issues. This is a rather old REV, How about you report your back trace to jira as per the instructions above. > > Thanks > -Josh > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Mar 25 12:42:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 14:42:50 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238006985.18397.1307291893@webmail.messagingengine.com> References: <1238006985.18397.1307291893@webmail.messagingengine.com> Message-ID: <10BEDCFA-A9C8-4157-A5C0-45CCA42199AB@freeswitch.org> btw you'll have to reinstall your phrase macros .... make vm-sync I think should do it if it doesn't let me know... we removed the 250ms sleeps and that was the problem which we fixed. /b On Mar 25, 2009, at 1:49 PM, Chris Fowler wrote: >>> First off what SVN rev? Remember when reporting issues try to >>> include all the information you can! > > Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647) Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com From mszlazak at aol.com Wed Mar 25 12:50:58 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 25 Mar 2009 15:50:58 -0400 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: References: Message-ID: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> I'd like to do something like "Make current" but for Windows because I'm finding bugs to report.? One was on how the dialplan's extensions are being parsed. Extensions in some cases like when doing originates with sched_api, loose their last character and I have to add a white space after to solve the problem. Another issue is that dialing one extension landed me into another totally different extension. I had to comment it out to target the right extension. Maybe the reporting bugs wiki needs updating for Windows users (experienced and inexperienced). Thanks. Mark. ? -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Wed, 25 Mar 2009 12:07 pm Subject: Re: [Freeswitch-users] Question about FAQ question On Mar 25, 2009, at 2:01 PM, Josh Forman wrote: > I'm running Freeswitch on Ubuntu 64bit Intrepid and on a svn rev > 12722, freeswitch would install and run fine but as soon as calls > started coming in it would have a segmentation fault. This is the > second svn snapshot I've had this happen on. Please review http://wiki.freeswitch.org/wiki/Reporting_Bugs Collect the info and report it on jira. If it still happens on SVN trunk as of NOW then please collect the back trace as per the reporting bugs guide and we'll investigate the issue. Also be aware if you aren't doing a "make current", you could have build skew which could be the whole problem all along. Incorrectly updating your system will result in all kinds of strange behaviors. > In the Freeswitch FAQs there is a question concerning segmentation > fault on ubuntu 64bit except they say it occurs on start. Their > solution is to recompile libedit which I plan to try regardless, but I > just wanted to know if the scenario they refer to is the same as what > I'm experiencing or was there some other problem where freeswitch > would segfault while initially loading. I don't think this is the problem you're having. > > Until then I'm still running rev 12289 on the system that is having > issues. This is a rather old REV, How about you report your back trace to jira as per the instructions above. > > Thanks > -Josh > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/46dc4a0c/attachment-0002.html From brian at freeswitch.org Wed Mar 25 12:55:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Mar 2009 14:55:14 -0500 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> References: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> Message-ID: <08EB85E2-4B2F-4A2D-99DB-29F0F753AB4A@freeswitch.org> If you svn up clean the solution and rebuild its the same thing.... Can you provide me a test case of this happening? /b On Mar 25, 2009, at 2:50 PM, mszlazak at aol.com wrote: > I'd like to do something like "Make current" but for Windows because > I'm finding bugs to report. > > One was on how the dialplan's extensions are being parsed. > Extensions in some cases like when doing originates with sched_api, > loose their last character and I have to add a white space after to > solve the problem. Another issue is that dialing one extension > landed me into another totally different extension. I had to comment > it out to target the right extension. > Maybe the reporting bugs wiki needs updating for Windows users > (experienced and inexperienced). > > Thanks. Mark. > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/df04f9b2/attachment-0002.html From msc at freeswitch.org Wed Mar 25 12:59:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 12:59:31 -0700 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> References: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> Message-ID: <87f2f3b90903251259w45ae04c0y5f81bfdebb131c54@mail.gmail.com> > Maybe the reporting bugs wiki needs updating for Windows users (experienced > and inexperienced). Quite possibly. The instructions are not explicit. I will add something for the Windows users that's a bit more specific. -MC From mszlazak at aol.com Wed Mar 25 13:37:50 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 25 Mar 2009 16:37:50 -0400 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: <87f2f3b90903251259w45ae04c0y5f81bfdebb131c54@mail.gmail.com> References: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> <87f2f3b90903251259w45ae04c0y5f81bfdebb131c54@mail.gmail.com> Message-ID: <8CB7BA0B282A3A3-424-7D@WEBMAIL-MB17.sysops.aol.com> Thanks MC, maybe a link to that "TortoiseSVN" would help for some in the Windows crowd. TortoiseSVN has a bunch of stuff in it but to make it simple, especially for doing updates to report bugs, then mentioning if doing just an "SVN update" will work before rebuild. Also, what to do if one gets an errror using SVN Update. I got one about not being able to open a file. So, I didn't know if any of the rest of the SVN update succeeded. I guess it wasn't a "clean" update. I didn't bother rebuilding afterwards since I also didn't know if it would work. There was to much to go through in TortoiseSVN documentation for the time I had so I didn't report the errors and left things for later when I would just download and install a newer version of FS. It sounds lazy but as an inexperienced user it's enough discouragment to let these things go. Anyway, just a bit more instructions might get more bugs reported. Thanks again. Mark. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Sent: Wed, 25 Mar 2009 12:59 pm Subject: Re: [Freeswitch-users] Question about FAQ question > Maybe the reporting bugs wiki needs updating for Windows users (experienced > and inexperienced). Quite possibly. The instructions are not explicit. I will add something for the Windows users that's a bit more specific. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/61610745/attachment-0002.html From msc at freeswitch.org Wed Mar 25 13:48:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 13:48:51 -0700 Subject: [Freeswitch-users] Question about FAQ question In-Reply-To: <8CB7BA0B282A3A3-424-7D@WEBMAIL-MB17.sysops.aol.com> References: <8CB7B9A26E82AF3-180-1FF5@WEBMAIL-MB17.sysops.aol.com> <87f2f3b90903251259w45ae04c0y5f81bfdebb131c54@mail.gmail.com> <8CB7BA0B282A3A3-424-7D@WEBMAIL-MB17.sysops.aol.com> Message-ID: <87f2f3b90903251348v1f47b8eau34fcb4d8c8d2759f@mail.gmail.com> 2009/3/25 : > Thanks MC, maybe a link to that "TortoiseSVN" would help for some in the > Windows crowd. Done! > > TortoiseSVN has a bunch of stuff in it but to make it simple, especially for > doing updates to report bugs, then mentioning if doing just an "SVN update" > will work before rebuild. Also, what to do if one gets an errror using SVN > Update. > I got one about not being able to open a file. So, I didn't know if any of > the rest of the SVN update succeeded. I guess it wasn't a "clean" update. > I didn't bother rebuilding afterwards since I also didn't know if it would > work. There was to much to go through in TortoiseSVN documentation for the > time I had so I didn't report the errors and left things for later when I > would just download and install a newer version of FS. > It sounds lazy but as an inexperienced user it's enough discouragment to let > these things go. > Anyway, just a bit more instructions might get more bugs reported. Understood. We can't anticipate every possible error and give documentation otherwise the Reporting Bugs page would be a lot larger than it is. In cases like this I recommend the panacea for all ills: Google. :) -MC From msc at freeswitch.org Wed Mar 25 13:53:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Mar 2009 13:53:02 -0700 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <49CA3978.7020704@gcdf.pl> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> <49CA2B59.6070708@gcdf.pl> <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> <49CA3978.7020704@gcdf.pl> Message-ID: <87f2f3b90903251353g1fe1e1d7i874e2593ed2c593c@mail.gmail.com> Hey, if you guys get this all figured out, tested, and working then please be sure to put it on the wiki. You could create a whole new page and then link to/from the mod_conference page. -MC On Wed, Mar 25, 2009 at 7:02 AM, Szymon Olko wrote: > Steven Ward pisze: >> Szymon, I want to provide a service wherein a user can reserve a >> teleconference room for a partiuclar time and control who can join the >> conference call (only those invited). ?I want to support several of >> such conference calls at any given time. >> >> I want callers who were not invited to a conference call, but who try to >> call in, to be notified they can't join. >> > Reserving conference means some storage of it. I think you have write some application to control it. > You can use store that in dialplan but it will be hard to make such dialplan. > There is pin code for conference so you can use it. > >> Also, it should be simple enough that the user just has to dial a number >> and be able to know what to do from there to join the conference. >> >> If the conference leader is late to the conference call and others try >> to call in (before the conference is created), the callers should be >> notified the conf leader hasn't started the conference yet. ?Perhaps >> they can be told to try later, or they can be put on hold. >> >> These are the kinds of things I need to think about implementing, and I >> do not know yet what other features would be useful. ?Thanks. > > I would make it like that. If you need to limit total number of conferences then you have to manage time of reservation. If you > don't need that then make one extension in which you do the following. > - get pincode from user. > - get admin_pincode from user. > - look in db for next available conference number > - store conference number and pincode in database. > - return that number to user. > > Make second extension where you do folowing: > - get conference number from user > - get pincode from user > - check in db if conference and pincode are the same as stored in db. > - check in database if admin is in conference. > > > Make third extension: > - get conference number from user > - get admin pincode from user > - check for matching in db > - transfer to conference > - store in db that admin is online > > > With this way users must get conference name and pincode to access. You can change it and in first extension add ability for user > to add others members with their personal number or something like that. > > Szymon Olko > >> On Wed, Mar 25, 2009 at 9:02 AM, Szymon Olko wrote: >> >> ? ? Steven Ward pisze: >> ? ? > If any can share some ideas, I'm looking at making conference >> ? ? > calls simple for the end-users of my FS system. >> ? ? > >> ? ? > Here are some issues I'm kicking around: >> ? ? > >> ? ? > 1. ?To create/join conferences - do you make a pre-defined list of >> ? ? > extensions, each of which would join the caller to a particular >> ? ? > conference room, or do you perhaps define one extension that will join >> ? ? > the caller to a dynamically-named conference room (perhaps after >> ? ? making >> ? ? > a choice in an IVR)? >> ? ? > >> ? ? > 2. ?How do you implement the idea of a conference leader - having a >> ? ? > user who must call into a conference first before anybody else can >> ? ? join? >> ? ? > >> ? ? > Thanks for any ideas. >> ? ? > >> ? ? Can you give more info what you plan to do? >> >> ? ? If you don't care for conferences name and they can be numbers then >> ? ? just make one extension and then use that extension or part of >> ? ? it to name conference. >> >> ? ? If you want to make conference leader then make 2 separate >> ? ? extensions, in second extension which is used for not leaders use that >> ? ? conference command. 'conference xxx bgdial '.In current >> ? ? SVN, bgdial does not create new conference it must exists to >> ? ? make that command. >> >> ? ? This can be make just by dial plan. If you plan to control FS from >> ? ? any language then you have much more abilities. >> >> ? ? > - SW >> ? ? > >> ? ? > >> ? ? > >> ? ? ------------------------------------------------------------------------ >> ? ? > >> ? ? > _______________________________________________ >> ? ? > Freeswitch-users mailing list >> ? ? > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steve.d.ward at gmail.com Wed Mar 25 14:03:23 2009 From: steve.d.ward at gmail.com (Steven Ward) Date: Wed, 25 Mar 2009 17:03:23 -0400 Subject: [Freeswitch-users] conference calling ideas In-Reply-To: <87f2f3b90903251353g1fe1e1d7i874e2593ed2c593c@mail.gmail.com> References: <4ea6e8f20903250544p48aedfe0laab191a8608628a7@mail.gmail.com> <49CA2B59.6070708@gcdf.pl> <4ea6e8f20903250637u3e350deam2ce83acc54eef11f@mail.gmail.com> <49CA3978.7020704@gcdf.pl> <87f2f3b90903251353g1fe1e1d7i874e2593ed2c593c@mail.gmail.com> Message-ID: <4ea6e8f20903251403k37ea8331y7087be3b59edc2e@mail.gmail.com> I need some time to work out my setup and explore some different options, but I'll be happy to get something together for the wiki on this as soon as I'm able. Thanks. On Wed, Mar 25, 2009 at 4:53 PM, Michael Collins wrote: > Hey, if you guys get this all figured out, tested, and working then > please be sure to put it on the wiki. You could create a whole new > page and then link to/from the mod_conference page. > -MC > > On Wed, Mar 25, 2009 at 7:02 AM, Szymon Olko wrote: > > Steven Ward pisze: > >> Szymon, I want to provide a service wherein a user can reserve a > >> teleconference room for a partiuclar time and control who can join the > >> conference call (only those invited). I want to support several of > >> such conference calls at any given time. > >> > >> I want callers who were not invited to a conference call, but who try to > >> call in, to be notified they can't join. > >> > > Reserving conference means some storage of it. I think you have write > some application to control it. > > You can use store that in dialplan but it will be hard to make such > dialplan. > > There is pin code for conference so you can use it. > > > >> Also, it should be simple enough that the user just has to dial a number > >> and be able to know what to do from there to join the conference. > >> > >> If the conference leader is late to the conference call and others try > >> to call in (before the conference is created), the callers should be > >> notified the conf leader hasn't started the conference yet. Perhaps > >> they can be told to try later, or they can be put on hold. > >> > >> These are the kinds of things I need to think about implementing, and I > >> do not know yet what other features would be useful. Thanks. > > > > I would make it like that. If you need to limit total number of > conferences then you have to manage time of reservation. If you > > don't need that then make one extension in which you do the following. > > - get pincode from user. > > - get admin_pincode from user. > > - look in db for next available conference number > > - store conference number and pincode in database. > > - return that number to user. > > > > Make second extension where you do folowing: > > - get conference number from user > > - get pincode from user > > - check in db if conference and pincode are the same as stored in db. > > - check in database if admin is in conference. > > > > > > Make third extension: > > - get conference number from user > > - get admin pincode from user > > - check for matching in db > > - transfer to conference > > - store in db that admin is online > > > > > > With this way users must get conference name and pincode to access. You > can change it and in first extension add ability for user > > to add others members with their personal number or something like that. > > > > Szymon Olko > > > >> On Wed, Mar 25, 2009 at 9:02 AM, Szymon Olko wrote: > >> > >> Steven Ward pisze: > >> > If any can share some ideas, I'm looking at making conference > >> > calls simple for the end-users of my FS system. > >> > > >> > Here are some issues I'm kicking around: > >> > > >> > 1. To create/join conferences - do you make a pre-defined list of > >> > extensions, each of which would join the caller to a particular > >> > conference room, or do you perhaps define one extension that will > join > >> > the caller to a dynamically-named conference room (perhaps after > >> making > >> > a choice in an IVR)? > >> > > >> > 2. How do you implement the idea of a conference leader - having > a > >> > user who must call into a conference first before anybody else can > >> join? > >> > > >> > Thanks for any ideas. > >> > > >> Can you give more info what you plan to do? > >> > >> If you don't care for conferences name and they can be numbers then > >> just make one extension and then use that extension or part of > >> it to name conference. > >> > >> If you want to make conference leader then make 2 separate > >> extensions, in second extension which is used for not leaders use > that > >> conference command. 'conference xxx bgdial '.In current > >> SVN, bgdial does not create new conference it must exists to > >> make that command. > >> > >> This can be make just by dial plan. If you plan to control FS from > >> any language then you have much more abilities. > >> > >> > - SW > >> > > >> > > >> > > >> > ------------------------------------------------------------------------ > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/b3b1940c/attachment-0002.html From emercado at rapidlink.com Wed Mar 25 13:01:17 2009 From: emercado at rapidlink.com (chevio) Date: Wed, 25 Mar 2009 13:01:17 -0700 (PDT) Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> Message-ID: <22709805.post@talk.nabble.com> How was this fixed ?, I am experiencing the same problem. Chevio Shelby Ramsey-2 wrote: > > Thanks for the help. That did the trick. > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Compile-Errors-...-tp22149989p22709805.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Wed Mar 25 14:39:21 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Mar 2009 17:39:21 -0400 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <22709805.post@talk.nabble.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> <22709805.post@talk.nabble.com> Message-ID: it was never really fixed as no one let me into their machine to troubleshoot. Mike On Mar 25, 2009, at 4:01 PM, chevio wrote: > > How was this fixed ?, I am experiencing the same problem. > > Chevio > > > Shelby Ramsey-2 wrote: >> >> Thanks for the help. That did the trick. >> >> SDR >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Compile-Errors-...-tp22149989p22709805.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From congxin.zhao at gmail.com Wed Mar 25 18:49:13 2009 From: congxin.zhao at gmail.com (congxin zhao) Date: Thu, 26 Mar 2009 09:49:13 +0800 Subject: [Freeswitch-users] help about the rtp port. In-Reply-To: <58A60265-2CBF-409A-8483-7371F82AD4B6@freeswitch.org> References: <58A60265-2CBF-409A-8483-7371F82AD4B6@freeswitch.org> Message-ID: Hi Brian, Thanks for you response. After doing research, the two endpoints(uac, uas) wants to speak with different media format, that cause the freeswitch reject the rtp flow. But could you give me an detailed explain on the meaning of nbound-bypass-media and inbound-proxy-media? Does nbound-bypass-media mean that rtp flow directly from endpoint uac to endpoin uas, and inbound-proxy-media mean that rtp flow from endpoint uacfirst arrive freeswitch, and freeswitch forward it to the endpoint uas? Thanks, -Congxin 2009/3/26 Brian West > Zero for the port means something please provide a complete sip trace. > /b > > On Mar 25, 2009, at 5:52 AM, congxin zhao wrote: > > Hi, > > I meet a issue about the rtp port. The media port value in the sdp of SIP > 200 OK which freeswitch internal sends to ua is always 0, so the media > forward is always wrong. What could cause this problem possibly? > > No matter I set > > or > > The problem always showes. > > Does the following rtp-ip and ext-rtp-ip set correct? In external.xml, I > set both rtp-ip and ext-rtp-ip as $${external_rtp_ip}, is that correct? I > am not clear about this two variables in the external.xml and internal.xml > > external.xml: > external.xml: > > internal.xml: > internal.xml: > > > Thanks, > -Congxin > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/be3c30a4/attachment-0002.html From rjcajax at gmail.com Wed Mar 25 19:13:16 2009 From: rjcajax at gmail.com (Robert Clayton) Date: Wed, 25 Mar 2009 22:13:16 -0400 Subject: [Freeswitch-users] Lua session:setInputCallback Message-ID: All, When using Lua InputCallback while streaming audio and collecting an undetermined number of digits (finished by #) it seem that returning false, break or stop all close the InputCallback ability. I do not see the difference between the three? My question is in order to collect multiple digits the streaming must continue or be paused. Is there a way to stop the recording in such a way that digits can continue to be collected other than using pause? Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090325/39a2932b/attachment-0002.html From mike at jerris.com Wed Mar 25 19:48:53 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Mar 2009 22:48:53 -0400 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <22709805.post@talk.nabble.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> <22709805.post@talk.nabble.com> Message-ID: <6A8AEA04-A2B1-46A7-9691-52D7AC791406@jerris.com> Thanks for access to your machine. The issue was that the odbc detection was trying to use odbc if either the libs or headers were found, not only if both were found. I fixed the detection to not try to use odbc if the headers were not installed. Installing unixODBC- devel package of course lets you install as normal, but we will not try to install automatically when we can't. Mike On Mar 25, 2009, at 4:01 PM, chevio wrote: > > How was this fixed ?, I am experiencing the same problem. > > Chevio > > > Shelby Ramsey-2 wrote: >> >> Thanks for the help. That did the trick. >> >> SDR >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Compile-Errors-...-tp22149989p22709805.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Wed Mar 25 21:39:31 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 26 Mar 2009 15:39:31 +1100 Subject: [Freeswitch-users] Multiple calls with PortAudio Message-ID: <20090326043931.GA6652@jdc.jasonjgw.net> This has occurred with a number of recent revisions. If anyone can reproduce it or suggest debugging steps, I'll gladly supply more information. Jira isn't convenient for me due to X issues at the moment, unless there's a way to interact with it other than via a Javascript-capable Web browser. Distribution: Debian Sid, now kernel 2.6.29 (same experience with 2.6.26). FreeSwitch revision 12701. Steps to reproduce Make a call with portAudio to a SIP endpoint (I haven't tried it with other endpoints): pa call Next, make another call: pa call Result: the party to the first call hears no music on hold, or possibly broken/intermittent music on hold, and the second call fails to go through. Where it hangs seems to vary, and sometimes it does in fact work. Often it hangs before it reaches the calling state, suggesting that there's some kind of locking issue involved. I'll perform further testing. Meanwhile, if anyone else can reproduce it, this may help. Note that the second call often doesn't return an error; it just hangs somewhere in the set-up process, e.g., init. From jason at jasonjgw.net Wed Mar 25 22:52:19 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 26 Mar 2009 16:52:19 +1100 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles In-Reply-To: <20090323235646.GA11567@jdc.jasonjgw.net> References: <20090323063100.GA5058@jdc.jasonjgw.net> <20090323235646.GA11567@jdc.jasonjgw.net> Message-ID: <20090326055219.GA7791@jdc.jasonjgw.net> Jason White wrote: > It looks like an operating system issue to me. Furthermore, the following message on linux-kernel appears relevant. http://linux.derkeiler.com/Mailing-Lists/Kernel/2005-03/3988.htm >From what I have been able to ascertain, Red Hat/Fedora kernels don't seem to give rise to this issue, whereas Debian kernels do. I haven't yet tested with a mainline kernel.org kernel. From jason at jasonjgw.net Wed Mar 25 23:25:10 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 26 Mar 2009 17:25:10 +1100 Subject: [Freeswitch-users] Intermittent startup failures with TLS-enabled profiles In-Reply-To: <20090326055219.GA7791@jdc.jasonjgw.net> References: <20090323063100.GA5058@jdc.jasonjgw.net> <20090323235646.GA11567@jdc.jasonjgw.net> <20090326055219.GA7791@jdc.jasonjgw.net> Message-ID: <20090326062510.GA8310@jdc.jasonjgw.net> I've read the ipv6(7) manual page now. Unfortunately echo 1 > /proc/sys/net/ipv6/bindv6only doesn't solve the problem as the manual page suggests it should: From moizchinoy at gmail.com Wed Mar 25 23:59:34 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 26 Mar 2009 10:59:34 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... Message-ID: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> Hi, I am having trouble compiling iksemel for google talk. There errors are in gnutls.h... I followed the instructions on: http://wiki.freeswitch.org/wiki/Dingaling http://wiki.freeswitch.org/wiki/Ixemel_MSVS_project_example GNUTLS VERSION is "2.7.3". Here are the erors: 1. Error 1 error C2061: syntax error : identifier 'gnutls_record_send' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 2. Error 2 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 3. Error 3 error C2059: syntax error : 'type' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 4. Error 4 error C2061: syntax error : identifier 'gnutls_record_recv' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 5. Error 5 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 6. Error 6 error C2059: syntax error : 'type' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 7. Error 7 error C2061: syntax error : identifier 'gnutls_record_set_max_size' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 8. Error 8 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 9. Error 9 error C2059: syntax error : 'type' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 10. Error 10 error C2143: syntax error : missing ')' before '*' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 11. Error 11 error C2143: syntax error : missing '{' before '*' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 12. Error 12 error C2059: syntax error : ')' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 13. Error 13 error C2143: syntax error : missing ')' before '*' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 14. Error 14 error C2143: syntax error : missing '{' before '*' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 15. Error 15 error C2059: syntax error : ')' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 16. Error 16 error C2146: syntax error : missing ')' before identifier 'push_func' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 17. Error 17 error C2081: 'gnutls_push_func' : name in formal parameter list illegal D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 18. Error 18 error C2061: syntax error : identifier 'push_func' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 19. Error 19 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 20. Error 20 error C2059: syntax error : ')' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 21. Error 21 error C2146: syntax error : missing ')' before identifier 'pull_func' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 22. Error 22 error C2081: 'gnutls_pull_func' : name in formal parameter list illegal D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 23. Error 23 error C2061: syntax error : identifier 'pull_func' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 24. Error 24 error C2059: syntax error : ';' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 25. Error 25 error C2059: syntax error : ')' D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 26. Error 26 error C2146: syntax error : missing ')' before identifier 'tls_push' d:\freeswitch-snapshot\libs\iksemel\src\stream.c 105 27. Error 27 error C2059: syntax error : ')' d:\freeswitch-snapshot\libs\iksemel\src\stream.c 105 28. Error 28 error C2146: syntax error : missing ')' before identifier 'tls_pull' d:\freeswitch-snapshot\libs\iksemel\src\stream.c 106 29. Error 29 error C2059: syntax error : ')' d:\freeswitch-snapshot\libs\iksemel\src\stream.c 106 -- Regards, Moiz Chinoy. From mike at jerris.com Thu Mar 26 01:24:18 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Mar 2009 04:24:18 -0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> Message-ID: <3BE666D9-3080-4FAD-A44B-78F230AB3EDB@jerris.com> There is no known working build on windows for the tls with freeswitch. We would be happy if someone would submit a full working build. Mike On Mar 26, 2009, at 2:59 AM, Moiz Chinoy wrote: > Hi, > > I am having trouble compiling iksemel for google talk. There errors > are in gnutls.h... > I followed the instructions on: > http://wiki.freeswitch.org/wiki/Dingaling > http://wiki.freeswitch.org/wiki/Ixemel_MSVS_project_example > > GNUTLS VERSION is "2.7.3". > > Here are the erors: > > 1. > Error 1 error C2061: syntax error : identifier > 'gnutls_record_send' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 > 2. > Error 2 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 > 3. > Error 3 error C2059: syntax error : 'type' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 454 > 4. > Error 4 error C2061: syntax error : identifier > 'gnutls_record_recv' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 > 5. > Error 5 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 > 6. > Error 6 error C2059: syntax error : 'type' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 456 > 7. > Error 7 error C2061: syntax error : identifier > 'gnutls_record_set_max_size' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 > 8. > Error 8 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 > 9. > Error 9 error C2059: syntax error : 'type' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 468 > 10. > Error 10 error C2143: syntax error : missing ')' before '*' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 > 11. > Error 11 error C2143: syntax error : missing '{' before '*' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 > 12. > Error 12 error C2059: syntax error : ')' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 877 > 13. > Error 13 error C2143: syntax error : missing ')' before '*' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 > 14. > Error 14 error C2143: syntax error : missing '{' before '*' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 > 15. > Error 15 error C2059: syntax error : ')' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 879 > 16. > Error 16 error C2146: syntax error : missing ')' before > identifier 'push_func' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 17. > Error 17 error C2081: 'gnutls_push_func' : name in formal > parameter list illegal > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 18. > Error 18 error C2061: syntax error : identifier 'push_func' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 19. > Error 19 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 20. > Error 20 error C2059: syntax error : ')' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 896 > 21. > Error 21 error C2146: syntax error : missing ')' before > identifier 'pull_func' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 22. > Error 22 error C2081: 'gnutls_pull_func' : name in formal > parameter list illegal > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 23. > Error 23 error C2061: syntax error : identifier 'pull_func' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 24. > Error 24 error C2059: syntax error : ';' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 25. > Error 25 error C2059: syntax error : ')' > D:\Downloads\gnutls-2.7.3\include\gnutls\gnutls.h 898 > 26. > Error 26 error C2146: syntax error : missing ')' before > identifier 'tls_push' > d:\freeswitch-snapshot\libs\iksemel\src\stream.c 105 > 27. > Error 27 error C2059: syntax error : ')' > d:\freeswitch-snapshot\libs\iksemel\src\stream.c 105 > 28. > Error 28 error C2146: syntax error : missing ')' before > identifier 'tls_pull' > d:\freeswitch-snapshot\libs\iksemel\src\stream.c 106 > 29. > Error 29 error C2059: syntax error : ')' > d:\freeswitch-snapshot\libs\iksemel\src\stream.c 106 > > -- > Regards, > Moiz Chinoy. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mattdfong at gmail.com Thu Mar 26 02:40:17 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 16:40:17 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> Message-ID: <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> Hi Anthony, So it's been 2 days since my last request, so I'm due for another one ;) It would be nice if there was a way to execute a script (lua) on fifo bridge. I currently rely on the channel_bridge event, but I'm worried that as my system scales, it would be better to fire a custom event. In non-fifo mode, I can do this with bridge_pre_execute_bleg_app/data, but this doesn't work with a fifo bridge. It doesn't really matter which channel it fires on fifo out or fifo in channel, anything is better than having to listen for a specific channel_bridge on a high-use FS installation. Is there anyway to get an api/script to fire on fifo bridge currently that I'm missing? Thanks! --matt 2009/3/23 Anthony Minessale > ok, > maybe after this i can have a day off ;) > > 2 variables added to latest trunk: > > "fifo_caller_consumer_import" > "fifo_consumer_caller_import" > > both work like the regular import but one is a list of vars to copy from > caller to consumer and one is a list to copy from consumer to caller. > > > 2009/3/23 Matthew Fong > >> Thanks Anthony, for creating the transfer_after_bridge feature for me. >> Your rapid actions, are the reason I'm positive I made the right decision >> switch to to FS. >> I got another challenge to throw your way. In the current fifo >> implementation, there's no way to identify which fifo consumer, consumes a >> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >> pass data between channels when a fifo bridge is created). I want to be able >> to transfer some caller information to the consumer channel on bridge, to >> populate an agent's screen. >> >> Under normal (non-fifo) circumstances, when a call is bridged, I've used >> the 'import' channel variable, so that onBridge, the aleg automatically gets >> populated with the bleg's 'import' field. However when fifo bridges, it does >> not recognize import. In other applications, I've gotten around this by >> using bridge_pre_execute_bleg_app/data to throw a custom event but with >> fifo, bridge_pre_execute also does not work. I've been looking at the >> fifo::info event, but again, there's no fifo_action that directly links >> caller variables and consumer variables together. >> >> For now at least, I can get around this by storing uuid information in my >> separate database, and looking up the other channel's information based >> on other_leg_unique_id, but it would be nice if I could directly see it when >> the channel is bridged. Anyway, great program, and I hope you consider >> implementing these features to make FS even better. Thanks. >> >> --matt >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/f7b81407/attachment-0002.html From asannucci at gmail.com Thu Mar 26 04:23:47 2009 From: asannucci at gmail.com (Andrea) Date: Thu, 26 Mar 2009 06:23:47 -0500 Subject: [Freeswitch-users] Error Compiling iksemel... References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> Message-ID: <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> Are you installed gnutls and gnutls-devel? Regards From moizchinoy at gmail.com Thu Mar 26 04:46:16 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 26 Mar 2009 15:46:16 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> Message-ID: <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> Only gnutls 2.7.3 Not gnutls-devel On Thu, Mar 26, 2009 at 3:23 PM, Andrea wrote: > Are you installed gnutls and gnutls-devel? > > Regards > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. From helmut.kuper at ewetel.de Thu Mar 26 07:12:13 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 26 Mar 2009 15:12:13 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions Message-ID: <49CB8D3D.7050202@ewetel.de> Hello, since a few days I observe a high CPU load of my FS server, but I have no idea what it could be. There are only a few sessions running and there is only a few log activity. 2 days ago I restarted FS, but no change. The top command shows this: top - 15:02:33 up 106 days, 30 min, 4 users, load average: 0.24, 0.35, 0.42 Tasks: 190 total, 1 running, 189 sleeping, 0 stopped, 0 zombie Cpu(s): 7.2%us, 12.2%sy, 0.0%ni, 80.0%id, 0.2%wa, 0.2%hi, 0.2%si, 0.0%st Mem: 4151776k total, 4003664k used, 148112k free, 414708k buffers Swap: 15623204k total, 88k used, 15623116k free, 2021412k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 14048 ippbx 20 0 1555m 1.2g 10m S 43 30.5 333:11.03 freeswitch 14049 ippbx 20 0 1555m 1.2g 10m S 0 30.5 5:06.88 freeswitch 14054 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:14.38 freeswitch 14055 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:58.50 freeswitch 14057 ippbx 20 0 1555m 1.2g 10m S 0 30.5 13:05.20 freeswitch 20511 ippbx 20 0 1555m 1.2g 10m S 0 30.5 0:00.14 freeswitch so only one process (PID: 14048) is causing that load. It's not the parent process (the initial FS startup process) as ps -elf shows: ippbx at ippbx-prod-node0:~/ippbx.prod$ ps -eLf | grep frees ippbx 14033 1 14033 0 28 Mar23 ? 00:00:01 bin/freeswitch -nc ippbx 14033 1 14034 0 28 Mar23 ? 00:00:08 bin/freeswitch -nc ippbx 14033 1 14035 0 28 Mar23 ? 00:03:39 bin/freeswitch -nc ippbx 14033 1 14036 0 28 Mar23 ? 00:00:07 bin/freeswitch -nc ippbx 14033 1 14037 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 14038 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 14039 0 28 Mar23 ? 00:00:02 bin/freeswitch -nc ippbx 14033 1 14042 0 28 Mar23 ? 00:03:41 bin/freeswitch -nc ippbx 14033 1 14043 0 28 Mar23 ? 00:00:03 bin/freeswitch -nc ippbx 14033 1 14044 0 28 Mar23 ? 00:00:01 bin/freeswitch -nc ippbx 14033 1 14045 0 28 Mar23 ? 00:00:25 bin/freeswitch -nc ippbx 14033 1 14046 0 28 Mar23 ? 00:01:20 bin/freeswitch -nc ippbx 14033 1 14047 0 28 Mar23 ? 00:05:32 bin/freeswitch -nc ippbx 14033 1 14048 7 28 Mar23 ? 05:33:35 bin/freeswitch -nc ippbx 14033 1 14049 0 28 Mar23 ? 00:05:07 bin/freeswitch -nc ippbx 14033 1 14050 0 28 Mar23 ? 00:01:01 bin/freeswitch -nc ippbx 14033 1 14051 0 28 Mar23 ? 00:25:43 bin/freeswitch -nc ippbx 14033 1 14052 0 28 Mar23 ? 00:00:01 bin/freeswitch -nc ippbx 14033 1 14054 0 28 Mar23 ? 00:04:14 bin/freeswitch -nc ippbx 14033 1 14055 0 28 Mar23 ? 00:04:58 bin/freeswitch -nc ippbx 14033 1 14056 0 28 Mar23 ? 00:06:23 bin/freeswitch -nc ippbx 14033 1 14057 0 28 Mar23 ? 00:13:05 bin/freeswitch -nc ippbx 14033 1 14058 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 14059 0 28 Mar23 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20518 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20519 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20521 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 14033 1 20522 0 28 15:02 ? 00:00:00 bin/freeswitch -nc ippbx 20526 19854 20526 0 1 15:03 pts/0 00:00:00 grep frees Doing a strace on PID 14048 prints tons of "epoll_wait(21, {}, 4, 0) = 0" lines on the screen, which eats all of my desktop pc's cpu power :/ So can a developer say what this is, or what and how should I debug to find out the cause of this? Can I shot it down via kill or "kill -9" without crashing FS totally? regards helmut From trevor at concipient.net Thu Mar 26 02:45:58 2009 From: trevor at concipient.net (Trevor Hammonds) Date: Thu, 26 Mar 2009 02:45:58 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid Message-ID: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> Has there been any progress getting FreeSWITCH to build on Ubuntu Intrepid without downgrading libtool? Thanks! Sincerely, Trevor Hammonds From Richard.Lamkin at mettoni.com Thu Mar 26 04:24:53 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 26 Mar 2009 11:24:53 -0000 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> Dear All, As a developer within a commercial organisation I would like to highlight that IRC access is blocked by my organisation. This is because it falls under the chat room category and is regarded as a security risk. Therefore is there any means of putting a digest of IRC traffic though the IRC node used for Freeswitch. I like many in the commercial world are barred by IT departmental polices from any chat room access. I feel I'm missing out on this useful stream on information. Another issue with any medium which is transitory is that I work in the UK and an therefore would not be privy to communications that occur outside my time zone. I do support efforts to put together a forum, which although less response than IRC is more permanent. I am subscribed to the User and Dev mail lists which I find are a very useful read. Regards Richard Lamkin Mettoni Group, UK ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/1bad5e3b/attachment-0002.html From frank at impactfax.com Thu Mar 26 07:19:31 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 26 Mar 2009 10:19:31 -0400 Subject: [Freeswitch-users] compile poblem - FIXED_POINT or FLOATING_POINT Message-ID: <1de601c9ae1d$e136cf40$33014c0a@ws4> I have this version running on fedora 8 right now. Compiled fine and is in production. version FreeSWITCH Version 1.0.trunk (10960) However, I was in the src directory and ran "make current" and after starting to compile it blew up with Making all in libspeex make[4]: Entering directory `/usr/src/freeswitch/libs/speex/libspeex' /bin/sh ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c -o nb_celp.lo nb_celp.c gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c nb_celp.c -fPIC -DPIC -o nb_celp.o In file included from modes.h:41, from nb_celp.h:39, from nb_celp.c:37: arch.h:65:2: error: #error You now need to define either FIXED_POINT or FLOATING_POINT make[4]: *** [nb_celp.lo] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/libs/speex/libspeex' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/libs/speex' make[2]: *** [all] Error 2 make[2]: Leaving directory `/usr/src/freeswitch/libs/speex' make[1]: *** [libs/speex/libspeex/libspeexdsp.la] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 ----- from modules.conf- loggers/mod_console loggers/mod_logfile loggers/mod_syslog applications/mod_commands applications/mod_conference applications/mod_dptools applications/mod_enum applications/mod_fifo applications/mod_voicemail applications/mod_limit applications/mod_expr applications/mod_esf applications/mod_fsv asr_tts/mod_flite asr_tts/mod_pocketsphinx asr_tts/mod_cepstral codecs/mod_g711 codecs/mod_g723_1 codecs/mod_amr codecs/mod_g729 codecs/mod_h26x codecs/mod_voipcodecs codecs/mod_ilbc codecs/mod_speex dialplans/mod_dialplan_xml dialplans/mod_dialplan_asterisk endpoints/mod_iax endpoints/mod_sofia event_handlers/mod_event_socket event_handlers/mod_cdr_csv formats/mod_native_file formats/mod_sndfile formats/mod_local_stream formats/mod_tone_stream languages/mod_spidermonkey languages/mod_spidermonkey_teletone languages/mod_spidermonkey_core_db languages/mod_spidermonkey_socket languages/mod_lua xml_int/mod_xml_rpc xml_int/mod_xml_curl xml_int/mod_xml_cdr say/mod_say_en any insight on what might be doing on with this compile where it previously compiled fine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/f4523900/attachment-0002.html From mrene_lists at avgs.ca Thu Mar 26 07:26:10 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Mar 2009 10:26:10 -0400 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CB8D3D.7050202@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> Message-ID: <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> gcore -o fs [pid here] gdb /path/to/fs core.file thread apply all bt then look for the thread and show me the backtrace. Math On 26-Mar-09, at 10:12 AM, Helmut Kuper wrote: > Hello, > > since a few days I observe a high CPU load of my FS server, but I have > no idea what it could be. There are only a few sessions running and > there is only a few log activity. 2 days ago I restarted FS, but no > change. The top command shows this: > > top - 15:02:33 up 106 days, 30 min, 4 users, load average: 0.24, > 0.35, > 0.42 > Tasks: 190 total, 1 running, 189 sleeping, 0 stopped, 0 zombie > Cpu(s): 7.2%us, 12.2%sy, 0.0%ni, 80.0%id, 0.2%wa, 0.2%hi, 0.2%si, > 0.0%st > Mem: 4151776k total, 4003664k used, 148112k free, 414708k > buffers > Swap: 15623204k total, 88k used, 15623116k free, 2021412k > cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > 14048 ippbx 20 0 1555m 1.2g 10m S 43 30.5 333:11.03 > freeswitch > 14049 ippbx 20 0 1555m 1.2g 10m S 0 30.5 5:06.88 > freeswitch > 14054 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:14.38 > freeswitch > 14055 ippbx 20 0 1555m 1.2g 10m S 0 30.5 4:58.50 > freeswitch > 14057 ippbx 20 0 1555m 1.2g 10m S 0 30.5 13:05.20 > freeswitch > 20511 ippbx 20 0 1555m 1.2g 10m S 0 30.5 0:00.14 > freeswitch > > > so only one process (PID: 14048) is causing that load. It's not the > parent process (the initial FS startup process) as ps -elf shows: > > ippbx at ippbx-prod-node0:~/ippbx.prod$ ps -eLf | grep frees > ippbx 14033 1 14033 0 28 Mar23 ? 00:00:01 > bin/freeswitch -nc > ippbx 14033 1 14034 0 28 Mar23 ? 00:00:08 > bin/freeswitch -nc > ippbx 14033 1 14035 0 28 Mar23 ? 00:03:39 > bin/freeswitch -nc > ippbx 14033 1 14036 0 28 Mar23 ? 00:00:07 > bin/freeswitch -nc > ippbx 14033 1 14037 0 28 Mar23 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 14038 0 28 Mar23 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 14039 0 28 Mar23 ? 00:00:02 > bin/freeswitch -nc > ippbx 14033 1 14042 0 28 Mar23 ? 00:03:41 > bin/freeswitch -nc > ippbx 14033 1 14043 0 28 Mar23 ? 00:00:03 > bin/freeswitch -nc > ippbx 14033 1 14044 0 28 Mar23 ? 00:00:01 > bin/freeswitch -nc > ippbx 14033 1 14045 0 28 Mar23 ? 00:00:25 > bin/freeswitch -nc > ippbx 14033 1 14046 0 28 Mar23 ? 00:01:20 > bin/freeswitch -nc > ippbx 14033 1 14047 0 28 Mar23 ? 00:05:32 > bin/freeswitch -nc > ippbx 14033 1 14048 7 28 Mar23 ? 05:33:35 > bin/freeswitch -nc > ippbx 14033 1 14049 0 28 Mar23 ? 00:05:07 > bin/freeswitch -nc > ippbx 14033 1 14050 0 28 Mar23 ? 00:01:01 > bin/freeswitch -nc > ippbx 14033 1 14051 0 28 Mar23 ? 00:25:43 > bin/freeswitch -nc > ippbx 14033 1 14052 0 28 Mar23 ? 00:00:01 > bin/freeswitch -nc > ippbx 14033 1 14054 0 28 Mar23 ? 00:04:14 > bin/freeswitch -nc > ippbx 14033 1 14055 0 28 Mar23 ? 00:04:58 > bin/freeswitch -nc > ippbx 14033 1 14056 0 28 Mar23 ? 00:06:23 > bin/freeswitch -nc > ippbx 14033 1 14057 0 28 Mar23 ? 00:13:05 > bin/freeswitch -nc > ippbx 14033 1 14058 0 28 Mar23 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 14059 0 28 Mar23 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 20518 0 28 15:02 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 20519 0 28 15:02 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 20521 0 28 15:02 ? 00:00:00 > bin/freeswitch -nc > ippbx 14033 1 20522 0 28 15:02 ? 00:00:00 > bin/freeswitch -nc > ippbx 20526 19854 20526 0 1 15:03 pts/0 00:00:00 grep frees > > Doing a strace on PID 14048 prints tons of "epoll_wait(21, {}, 4, > 0) = 0" lines on the screen, which eats all of my > desktop > pc's cpu power :/ > > So can a developer say what this is, or what and how should I debug to > find out the cause of this? > Can I shot it down via kill or "kill -9" without crashing FS totally? > > regards > helmut > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mrene_lists at avgs.ca Thu Mar 26 07:26:57 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Mar 2009 10:26:57 -0400 Subject: [Freeswitch-users] compile poblem - FIXED_POINT or FLOATING_POINT In-Reply-To: <1de601c9ae1d$e136cf40$33014c0a@ws4> References: <1de601c9ae1d$e136cf40$33014c0a@ws4> Message-ID: <8E511016-5BC8-490A-9672-F1D53C580645@avgs.ca> make speex-reconf On 26-Mar-09, at 10:19 AM, Frank @ Impact wrote: > I have this version running on fedora 8 right now. Compiled fine > and is in production. > version > FreeSWITCH Version 1.0.trunk (10960) > > However, I was in the src directory and ran ?make current? and after > starting to compile it blew up with > > Making all in libspeex > make[4]: Entering directory `/usr/src/freeswitch/libs/speex/libspeex' > /bin/sh ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. > -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP > -MF .deps/nb_celp.Tpo -c -onb_celp.lo nb_celp.c > gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 - > MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c nb_celp.c -fPIC - > DPIC -o nb_celp.o > In file included from modes.h:41, > from nb_celp.h:39, > from nb_celp.c:37: > arch.h:65:2: error: #error You now need to define either FIXED_POINT > or FLOATING_POINT > make[4]: *** [nb_celp.lo] Error 1 > make[4]: Leaving directory `/usr/src/freeswitch/libs/speex/libspeex' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/src/freeswitch/libs/speex' > make[2]: *** [all] Error 2 > make[2]: Leaving directory `/usr/src/freeswitch/libs/speex' > make[1]: *** [libs/speex/libspeex/libspeexdsp.la] Error 2 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 > ----- > from modules.conf? > > loggers/mod_console > loggers/mod_logfile > loggers/mod_syslog > applications/mod_commands > applications/mod_conference > applications/mod_dptools > applications/mod_enum > applications/mod_fifo > applications/mod_voicemail > applications/mod_limit > applications/mod_expr > applications/mod_esf > applications/mod_fsv > asr_tts/mod_flite > asr_tts/mod_pocketsphinx > asr_tts/mod_cepstral > codecs/mod_g711 > codecs/mod_g723_1 > codecs/mod_amr > codecs/mod_g729 > codecs/mod_h26x > codecs/mod_voipcodecs > codecs/mod_ilbc > codecs/mod_speex > dialplans/mod_dialplan_xml > dialplans/mod_dialplan_asterisk > endpoints/mod_iax > endpoints/mod_sofia > event_handlers/mod_event_socket > event_handlers/mod_cdr_csv > formats/mod_native_file > formats/mod_sndfile > formats/mod_local_stream > formats/mod_tone_stream > languages/mod_spidermonkey > languages/mod_spidermonkey_teletone > languages/mod_spidermonkey_core_db > languages/mod_spidermonkey_socket > languages/mod_lua > xml_int/mod_xml_rpc > xml_int/mod_xml_curl > xml_int/mod_xml_cdr > say/mod_say_en > > > any insight on what might be doing on with this compile where it > previously compiled fine. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/f9980dbd/attachment-0002.html From telles-listas at devel-it.com.br Thu Mar 26 07:27:33 2009 From: telles-listas at devel-it.com.br (Rodrigo P. Telles) Date: Thu, 26 Mar 2009 11:27:33 -0300 Subject: [Freeswitch-users] Action and Anti-Action Message-ID: <49CB90D5.4090901@devel-it.com.br> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/421916b3/attachment-0002.html From brian at freeswitch.org Thu Mar 26 07:32:32 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 09:32:32 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> Message-ID: <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> http://cgiirc.freeswitch.org/ I'm assume the web isn't blocked? /b On Mar 26, 2009, at 6:24 AM, Richard Lamkin wrote: > Dear All, > > As a developer within a commercial organisation I would like to > highlight that IRC access is blocked by my organisation. This is > because it falls under the chat room category and is regarded as a > security risk. > > Therefore is there any means of putting a digest of IRC traffic > though the IRC node used for Freeswitch. I like many in the > commercial world are barred by IT departmental polices from any chat > room access. I feel I?m missing out on this useful stream on > information. Another issue with any medium which is transitory is > that I work in the UK and an therefore would not be privy to > communications that occur outside my time zone. > > I do support efforts to put together a forum, which although less > response than IRC is more permanent. > I am subscribed to the User and Dev mail lists which I find are a > very useful read. > > Regards > > Richard Lamkin > Mettoni Group, UK > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/a3ecb8a5/attachment-0002.html From frank at impactfax.com Thu Mar 26 07:39:46 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 26 Mar 2009 10:39:46 -0400 Subject: [Freeswitch-users] compile poblem - FIXED_POINT orFLOATING_POINT In-Reply-To: <8E511016-5BC8-490A-9672-F1D53C580645@avgs.ca> Message-ID: <1e1801c9ae20$b5b7adf0$33014c0a@ws4> Thanks. That did it. But I no longer could make mod_ilbc (and mod_flite). I had to comment it out of modules.conf because it would error with this making all mod_ilbc make[6]: *** No targets specified and no makefile found. Stop. make[5]: *** [../../../../libs/ilbc/src/libilbc.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_ilbc-all] Error 1 make[2]: *** [all-recursive] Error 1 make ilbc-reconf did not work -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Thursday, March 26, 2009 10:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] compile poblem - FIXED_POINT orFLOATING_POINT make speex-reconf On 26-Mar-09, at 10:19 AM, Frank @ Impact wrote: I have this version running on fedora 8 right now. Compiled fine and is in production. version FreeSWITCH Version 1.0.trunk (10960) However, I was in the src directory and ran "make current" and after starting to compile it blew up with Making all in libspeex make[4]: Entering directory `/usr/src/freeswitch/libs/speex/libspeex' /bin/sh ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c -onb_celp.lo nb_celp.c gcc -DHAVE_CONFIG_H -I. -I.. -I../include -I../include -I.. -g -O2 -MT nb_celp.lo -MD -MP -MF .deps/nb_celp.Tpo -c nb_celp.c -fPIC -DPIC -o nb_celp.o In file included from modes.h:41, from nb_celp.h:39, from nb_celp.c:37: arch.h:65:2: error: #error You now need to define either FIXED_POINT or FLOATING_POINT make[4]: *** [nb_celp.lo] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/libs/speex/libspeex' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/libs/speex' make[2]: *** [all] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/073efed4/attachment-0002.html From brian at freeswitch.org Thu Mar 26 07:44:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 09:44:42 -0500 Subject: [Freeswitch-users] compile poblem - FIXED_POINT orFLOATING_POINT In-Reply-To: <1e1801c9ae20$b5b7adf0$33014c0a@ws4> References: <1e1801c9ae20$b5b7adf0$33014c0a@ws4> Message-ID: <3797D5FE-C2D8-4238-A360-C170932D1D3E@freeswitch.org> how about you just rebootstrap it all.... /b On Mar 26, 2009, at 9:39 AM, Frank @ Impact wrote: > Thanks. That did it. > But I no longer could make mod_ilbc (and mod_flite). I had to > comment it out of modules.conf because it would error with this > > making all mod_ilbc > make[6]: *** No targets specified and no makefile found. Stop. > make[5]: *** [../../../../libs/ilbc/src/libilbc.la] Error 2 > make[4]: *** [all] Error 1 > make[3]: *** [mod_ilbc-all] Error 1 > make[2]: *** [all-recursive] Error 1 > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/98ce3a3d/attachment-0002.html From anthony.minessale at gmail.com Thu Mar 26 07:56:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 09:56:22 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> Message-ID: <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> The guy started a forum almost a month ago and as you can see nobody knows the url and it has no posts. http://freeswitch411.info/forum/ This is one of the problems I was worried about when endorsing a forum. 2009/3/26 Richard Lamkin > Dear All, > > > > As a developer within a commercial organisation I would like to highlight > that IRC access is blocked by my organisation. This is because it falls > under the chat room category and is regarded as a security risk. > > > > Therefore is there any means of putting a digest of IRC traffic though the > IRC node used for Freeswitch. I like many in the commercial world are > barred by IT departmental polices from any chat room access. I feel I?m > missing out on this useful stream on information. Another issue with any > medium which is transitory is that I work in the UK and an therefore would > not be privy to communications that occur outside my time zone. > > > > I do support efforts to put together a forum, which although less response > than IRC is more permanent. > > I am subscribed to the User and Dev mail lists which I find are a very > useful read. > > > > Regards > > > > Richard Lamkin > > Mettoni Group, UK > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/a0fb0f6c/attachment-0002.html From anthony.minessale at gmail.com Thu Mar 26 07:59:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 09:59:05 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> Message-ID: <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> We do not support ubuntu interpid, it has at least 3 known fatal issues not experienced by all but nonetheless enough to make us unwilling to support it. It's "use at your own risk" or use the stable branch "hardy" for any support. On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds wrote: > Has there been any progress getting FreeSWITCH to build on Ubuntu > Intrepid without downgrading libtool? > > Thanks! > > Sincerely, > Trevor Hammonds > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/b4cec9c4/attachment-0002.html From helmut.kuper at ewetel.de Thu Mar 26 08:01:53 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 26 Mar 2009 16:01:53 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> Message-ID: <49CB98E1.8080705@ewetel.de> Hi Mathieu, thx for the help :) The epoll_wait function on PID 14048 listens on fd 21 which points to "/anon_inode:[eventpoll]" On 26.03.2009 15:26, Mathieu Rene wrote: > thread apply all bt Here the output: Core was generated by `/opt/app/voip/ippbx.prod/bin/freeswitch'. [New process 14034] [New process 14035] [New process 14036] [New process 14037] [New process 14038] [New process 14039] [New process 14042] [New process 14043] [New process 14044] [New process 14045] [New process 14046] [New process 14047] [New process 14048] [New process 14049] [New process 14050] [New process 14051] [New process 14052] [New process 14054] [New process 14055] [New process 14056] [New process 14057] [New process 14058] [New process 14059] [New process 14033] #0 0xb7ee6410 in __kernel_vsyscall () (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) (gdb) thread apply all bt Thread 24 (process 14033): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7d9e334 in switch_core_runtime_loop (bg=1) at src/switch_core.c:656 #5 0x0804a459 in main (argc=2, argv=0xbf9d27a4) at src/switch.c:666 Thread 23 (process 14059): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c41c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb36ef7e1 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_xml_rpc.so #3 0xb36e2612 in ChanSwitchAccept (chanSwitchP=0x81cce68, channelPP=0xab1f00e0, channelInfoPP=0xab1f00dc, errorP=0xab1f00e4) at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 #4 0xb36ee351 in ServerRun (serverP=0xb372092c) at ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 #5 0xb36df892 in mod_xml_rpc_runtime () at mod_xml_rpc.c:867 #6 0xb7da8473 in switch_loadable_module_exec (thread=0x80c38a8, obj=0x80c3698) at src/switch_loadable_module.c:94 #7 0xb7dfeeb6 in dummy_worker (opaque=0x80c38a8) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 22 (process 14058): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7d00bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7dfdbcd in apr_socket_accept (new=0xab9f134c, sock=0x82639e8, connection_context=0x66a76500) at network_io/unix/sockets.c:187 #3 0xb7d7cf4b in switch_socket_accept (new_sock=0xab9f134c, sock=0x82639e8, pool=0x66a76500) at src/switch_apr.c:664 #4 0xb328ff2c in mod_event_socket_runtime () at mod_event_socket.c:2293 #5 0xb7da8473 in switch_loadable_module_exec (thread=0x80c3640, obj=0x80c3430) at src/switch_loadable_module.c:94 #6 0xb7dfeeb6 in dummy_worker (opaque=0x80c3640) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 21 (process 14057): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=1000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7de0c25 in softtimer_runtime () at src/switch_time.c:459 #5 0xb7da8473 in switch_loadable_module_exec (thread=0x80c33d8, obj=0x80c31c8) at src/switch_loadable_module.c:94 #6 0xb7dfeeb6 in dummy_worker (opaque=0x80c33d8) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 20 (process 14056): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at locks/unix/thread_cond.c:68 #3 0xb7d7d8b4 in switch_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at src/switch_apr.c:359 #4 0xb7ddf2c8 in timer_next (timer=0xaccae0f8) at src/switch_time.c:335 #5 0xb7d8ad7c in switch_core_timer_next (timer=0x26f180a8) at src/switch_core_timer.c:76 ---Type to continue, or q to quit--- #6 0xad4e67dc in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #7 0xb7dfeeb6 in dummy_worker (opaque=0xb272b2e0) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 19 (process 14055): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at locks/unix/thread_cond.c:68 #3 0xb7d7d8b4 in switch_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at src/switch_apr.c:359 #4 0xb7ddf2c8 in timer_next (timer=0xad4af0f8) at src/switch_time.c:335 #5 0xb7d8ad7c in switch_core_timer_next (timer=0x26f180a5) at src/switch_core_timer.c:76 #6 0xad4e67dc in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #7 0xb7dfeeb6 in dummy_worker (opaque=0xb27272d0) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 18 (process 14054): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at locks/unix/thread_cond.c:68 #3 0xb7d7d8b4 in switch_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at src/switch_apr.c:359 #4 0xb7ddf2c8 in timer_next (timer=0xadf700f8) at src/switch_time.c:335 #5 0xb7d8ad7c in switch_core_timer_next (timer=0x26f180a6) at src/switch_core_timer.c:76 #6 0xad4e67dc in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #7 0xb7dfeeb6 in dummy_worker (opaque=0xb27232c0) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 17 (process 14052): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xaedcce06 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so #5 0xb7dfeeb6 in dummy_worker (opaque=0xaea4cb90) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 16 (process 14051): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb315892f in sofia_presence_event_thread_run (thread=0x80f1bb0, obj=0x0) at sofia_presence.c:664 #5 0xb7dfeeb6 in dummy_worker (opaque=0x80f1bb0) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 15 (process 14050): #0 0xb7ee6410 in __kernel_vsyscall () ---Type to continue, or q to quit--- #1 0xb7c41c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb280bf00 in wanpipe_wait (zchan=0x8146fc8, flags=0xafe0ef8c, to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 #3 0xafeecb17 in zap_channel_wait (zchan=0x64, flags=0xafe0ef8c, to=100) at src/zap_io.c:1488 #4 0xafebfb54 in zap_isdn_run (me=0x8112ab0, obj=0x811ff58) at src/ozmod/ozmod_isdn/ozmod_isdn.c:1736 #5 0xafef747a in thread_launch (args=0x8112ab0) at src/zap_threadmutex.c:74 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 14 (process 14049): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb313e306 in sofia_profile_worker_thread_run (thread=0x81004c8, obj=0x80ffb38) at sofia.c:656 #5 0xb7dfeeb6 in dummy_worker (opaque=0x81004c8) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 13 (process 14048): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c4c676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb321998d in su_epoll_port_wait_events (self=0x8107f70, tout=1000) at su_epoll_port.c:495 #3 0xb3220218 in su_base_port_run (self=0x8107f70) at su_base_port.c:349 #4 0xb321af8f in su_port_run (self=0x8107f70) at su_port.h:326 #5 0xb321af6c in su_root_run (self=0x8107770) at su_root.c:819 #6 0xb320af0a in su_pthread_port_clone_main (varg=0xb16fd098) at su_pthread_port.c:324 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 12 (process 14047): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c4c676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb321998d in su_epoll_port_wait_events (self=0x80fbaa8, tout=1000) at su_epoll_port.c:495 #3 0xb3220369 in su_base_port_step (self=0x80fbaa8, tout=1000) at su_base_port.c:467 #4 0xb321b0b5 in su_port_step (self=0x80fbaa8, tout=1000) at su_port.h:340 #5 0xb321b085 in su_root_step (self=0x80fba70, tout=1000) at su_root.c:858 #6 0xb3148442 in sofia_profile_thread_run (thread=0x81003e8, obj=0x80ffb38) at sofia.c:831 #7 0xb7dfeeb6 in dummy_worker (opaque=0x81003e8) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 11 (process 14046): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb313e306 in sofia_profile_worker_thread_run (thread=0x80f5030, obj=0x80f4280) at sofia.c:656 #5 0xb7dfeeb6 in dummy_worker (opaque=0x80f5030) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- Thread 10 (process 14045): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c4c676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb321998d in su_epoll_port_wait_events (self=0x80fd1a0, tout=1000) at su_epoll_port.c:495 #3 0xb3220218 in su_base_port_run (self=0x80fd1a0) at su_base_port.c:349 #4 0xb321af8f in su_port_run (self=0x80fd1a0) at su_port.h:326 #5 0xb321af6c in su_root_run (self=0x80fef98) at su_root.c:819 #6 0xb320af0a in su_pthread_port_clone_main (varg=0xb30ac098) at su_pthread_port.c:324 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 9 (process 14044): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c4c676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb321998d in su_epoll_port_wait_events (self=0x80e0e80, tout=1000) at su_epoll_port.c:495 #3 0xb3220369 in su_base_port_step (self=0x80e0e80, tout=1000) at su_base_port.c:467 #4 0xb321b0b5 in su_port_step (self=0x80e0e80, tout=1000) at su_port.h:340 #5 0xb321b085 in su_root_step (self=0x80f8528, tout=1000) at su_root.c:858 #6 0xb3148442 in sofia_profile_thread_run (thread=0x80f4f50, obj=0x80f4280) at sofia.c:831 #7 0xb7dfeeb6 in dummy_worker (opaque=0x80f4f50) at threadproc/unix/thread.c:138 #8 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 8 (process 14043): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e00a59 in apr_sleep (t=500000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7da20a4 in switch_scheduler_task_thread (thread=0x80be590, obj=0x0) at src/switch_scheduler.c:171 #5 0xb7dfeeb6 in dummy_worker (opaque=0x80be590) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 7 (process 14042): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at locks/unix/thread_cond.c:68 #3 0xb7d7d8b4 in switch_thread_cond_wait (cond=0x80c4760, mutex=0x80c4730) at src/switch_apr.c:359 #4 0xb7ddf4d6 in switch_cond_next () at src/switch_time.c:203 #5 0xb7d91b5b in switch_core_sql_thread (thread=0xb3782ae8, obj=0x0) at src/switch_core_sqldb.c:220 #6 0xb7dfeeb6 in dummy_worker (opaque=0xb3782ae8) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 6 (process 14039): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0xb70ddc60, mutex=0xb70ddc30) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0xb70ddc00, data=0xb4fe93a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0xb70ddc00, data=0xb4fe93a8) at src/switch_apr.c:879 #5 0xb7dd7bed in log_thread (t=0xb504bae0, obj=0x0) at src/switch_log.c:213 ---Type to continue, or q to quit--- #6 0xb7dfeeb6 in dummy_worker (opaque=0xb504bae0) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 5 (process 14038): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0xb7140b38, mutex=0xb7140b08) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0xb7140ad8, data=0xb58783a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0xb7140ad8, data=0xb58783a8) at src/switch_apr.c:879 #5 0xb7dae2d0 in switch_event_thread (thread=0x8068228, obj=0xb7140ad8) at src/switch_event.c:284 #6 0xb7dfeeb6 in dummy_worker (opaque=0x8068228) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 4 (process 14037): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0xb71a3b38, mutex=0xb71a3b08) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0xb71a3ad8, data=0xb60793a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0xb71a3ad8, data=0xb60793a8) at src/switch_apr.c:879 #5 0xb7dae2d0 in switch_event_thread (thread=0x8068208, obj=0xb71a3ad8) at src/switch_event.c:284 #6 0xb7dfeeb6 in dummy_worker (opaque=0x8068208) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 3 (process 14036): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0x805e548, mutex=0x805e518) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0x805e4e8, data=0xb687a3a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0x805e4e8, data=0xb687a3a8) at src/switch_apr.c:879 #5 0xb7dae2d0 in switch_event_thread (thread=0x80681e8, obj=0x805e4e8) at src/switch_event.c:284 #6 0xb7dfeeb6 in dummy_worker (opaque=0x80681e8) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 2 (process 14035): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7cfdaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7df8f6a in apr_thread_cond_wait (cond=0xb70ddb38, mutex=0xb70ddb08) at locks/unix/thread_cond.c:68 #3 0xb7defcfc in apr_queue_pop (queue=0xb70ddad8, data=0xb707b3a8) at misc/apr_queue.c:276 #4 0xb7d7c884 in switch_queue_pop (queue=0xb70ddad8, data=0xb707b3a8) at src/switch_apr.c:879 #5 0xb7daf069 in switch_event_dispatch_thread (thread=0x80681c8, obj=0xb70ddad8) at src/switch_event.c:241 #6 0xb7dfeeb6 in dummy_worker (opaque=0x80681c8) at threadproc/unix/thread.c:138 #7 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 1 (process 14034): #0 0xb7ee6410 in __kernel_vsyscall () #1 0xb7c44881 in select () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- #2 0xb7e00a59 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7ddf09e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7d8d52b in pool_thread (thread=0xb79d6da8, obj=0x0) at src/switch_core_memory.c:423 #5 0xb7dfeeb6 in dummy_worker (opaque=0xb79d6da8) at threadproc/unix/thread.c:138 #6 0xb7cf94fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c4be5e in clone () from /lib/tls/i686/cmov/libc.so.6 From anthony.minessale at gmail.com Thu Mar 26 08:07:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 10:07:29 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> Message-ID: <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> I'll fire 2 custom events when the call is bridged one for the consumer and one for the caller events plain custom fifo::info pull out FIFO-Name header and find the desired fifo pull out FIFO-Action header and look for bridge-consumer or bridge-caller depending on what you want to see data from. in latest trunk 2009/3/26 Matthew Fong > Hi Anthony, > So it's been 2 days since my last request, so I'm due for another one ;) > > It would be nice if there was a way to execute a script (lua) on fifo > bridge. I currently rely on the channel_bridge event, but I'm worried that > as my system scales, it would be better to fire a custom event. In non-fifo > mode, I can do this with bridge_pre_execute_bleg_app/data, but this > doesn't work with a fifo bridge. It doesn't really matter which channel it > fires on fifo out or fifo in channel, anything is better than having to > listen for a specific channel_bridge on a high-use FS installation. > > Is there anyway to get an api/script to fire on fifo bridge currently that > I'm missing? Thanks! > > --matt > > 2009/3/23 Anthony Minessale > > ok, >> maybe after this i can have a day off ;) >> >> 2 variables added to latest trunk: >> >> "fifo_caller_consumer_import" >> "fifo_consumer_caller_import" >> >> both work like the regular import but one is a list of vars to copy from >> caller to consumer and one is a list to copy from consumer to caller. >> >> >> 2009/3/23 Matthew Fong >> >>> Thanks Anthony, for creating the transfer_after_bridge feature for me. >>> Your rapid actions, are the reason I'm positive I made the right decision >>> switch to to FS. >>> I got another challenge to throw your way. In the current fifo >>> implementation, there's no way to identify which fifo consumer, consumes a >>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>> pass data between channels when a fifo bridge is created). I want to be able >>> to transfer some caller information to the consumer channel on bridge, to >>> populate an agent's screen. >>> >>> Under normal (non-fifo) circumstances, when a call is bridged, I've used >>> the 'import' channel variable, so that onBridge, the aleg automatically gets >>> populated with the bleg's 'import' field. However when fifo bridges, it does >>> not recognize import. In other applications, I've gotten around this by >>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>> fifo, bridge_pre_execute also does not work. I've been looking at the >>> fifo::info event, but again, there's no fifo_action that directly links >>> caller variables and consumer variables together. >>> >>> For now at least, I can get around this by storing uuid information in my >>> separate database, and looking up the other channel's information based >>> on other_leg_unique_id, but it would be nice if I could directly see it when >>> the channel is bridged. Anyway, great program, and I hope you consider >>> implementing these features to make FS even better. Thanks. >>> >>> --matt >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/343d3bb4/attachment-0002.html From brian at freeswitch.org Thu Mar 26 08:07:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 10:07:56 -0500 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CB98E1.8080705@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> Message-ID: Before we go any further... what SVN rev are you on? And by heavy load what does your load average say? /b On Mar 26, 2009, at 10:01 AM, Helmut Kuper wrote: > Hi Mathieu, > > thx for the help :) > > The epoll_wait function on PID 14048 listens on fd 21 which points to > "/anon_inode:[eventpoll]" > > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/9de4eca2/attachment-0002.html From helmut.kuper at ewetel.de Thu Mar 26 08:12:08 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 26 Mar 2009 16:12:08 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CB98E1.8080705@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> Message-ID: <49CB9B48.5030603@ewetel.de> Hi Mathieu I straced process 14047 as well and I found that it also epolls a anon_inode. But the process' epoll call lokk slightly different that 14048's: epoll_wait(13, {}, 4, 1000) = 0 Here a timeout is given. 14048 is not using a timeout (0). Maybe this helps you ... regards Helmut On 26.03.2009 16:01, Helmut Kuper wrote: > Hi Mathieu, > > thx for the help :) > > The epoll_wait function on PID 14048 listens on fd 21 which points to > "/anon_inode:[eventpoll]" From mike at jerris.com Thu Mar 26 08:16:07 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Mar 2009 11:16:07 -0400 Subject: [Freeswitch-users] Action and Anti-Action In-Reply-To: <49CB90D5.4090901@devel-it.com.br> References: <49CB90D5.4090901@devel-it.com.br> Message-ID: Actions are all run AFTER all conditions are parsed so the nated var is not set yet. you can do a single condition in this case, and set nated for use elsewhere if you need it in the actions. Mike On Mar 26, 2009, at 10:27 AM, Rodrigo P. Telles wrote: > Hi Guys, > > I'm trying to do some string matching against a created var and > looks like I am doing something wrong but I can't find whats it. > > I'm wrote an extension just for tests purposes on dialplan/ > default.xml: > > > > > > > > > > > > > > Using two SIP extensions (1000 and 1001) behind NAT and I expected > too see "Action=1" on the logs/console, but I'm seeing "Anti- > Action=1". > > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x > Execute set(nated=${cond(${network_addr} != ${sip_contact_host} ? > 1 : 0)}) > 2009-03-26 11:02:57 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1001 at x.x.x.x Expanded > String set(nated=1) > 2009-03-26 11:02:57 [DEBUG] mod_dptools.c:711 set_function() sofia/internal/1001 at x.x.x.x > SET [nated]=[1] > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x > Execute log(Anti-Action=${nated}) > 2009-03-26 11:02:57 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1001 at x.x.x.x Expanded > String log(Anti-Action=1) > 2009-03-26 11:02:57 [DEBUG] mod_dptools.c:895 log_function() Anti- > Action=1 > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x > Execute set(dialed_extension=1000) > .... > > I really appreciate any inputs. > I'm using FS 1.0.3 stable. > > regards, > Rodrigo Telles > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/7af6dc65/attachment-0002.html From mattdfong at gmail.com Thu Mar 26 08:20:09 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 22:20:09 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> Message-ID: <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> Thanks of course! But, is there any chance of firing an app? Firing an app on bridge gives the programmer more control, rather than just listening for fifo::info custom events. I find that lua running as a FS app can update my database like 10x faster than reading event_socket thru Rails/Telegraph...plus, I trust your coding much more than that of your Rail's development counterparts. :) with the custom event you are firing, you should be sure to import the variables first, then fire the event :) You rock Mr. Minessale --matt 2009/3/26 Anthony Minessale > I'll fire 2 custom events when the call is bridged one for the consumer and > one for the caller > > events plain custom fifo::info > > pull out FIFO-Name header and find the desired fifo > pull out FIFO-Action header and look for bridge-consumer or bridge-caller > depending on what you want to see data from. > > in latest trunk > > 2009/3/26 Matthew Fong > > Hi Anthony, >> So it's been 2 days since my last request, so I'm due for another one ;) >> >> It would be nice if there was a way to execute a script (lua) on fifo >> bridge. I currently rely on the channel_bridge event, but I'm worried that >> as my system scales, it would be better to fire a custom event. In non-fifo >> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >> doesn't work with a fifo bridge. It doesn't really matter which channel it >> fires on fifo out or fifo in channel, anything is better than having to >> listen for a specific channel_bridge on a high-use FS installation. >> >> Is there anyway to get an api/script to fire on fifo bridge currently that >> I'm missing? Thanks! >> >> --matt >> >> 2009/3/23 Anthony Minessale >> >> ok, >>> maybe after this i can have a day off ;) >>> >>> 2 variables added to latest trunk: >>> >>> "fifo_caller_consumer_import" >>> "fifo_consumer_caller_import" >>> >>> both work like the regular import but one is a list of vars to copy from >>> caller to consumer and one is a list to copy from consumer to caller. >>> >>> >>> 2009/3/23 Matthew Fong >>> >>>> Thanks Anthony, for creating the transfer_after_bridge feature for me. >>>> Your rapid actions, are the reason I'm positive I made the right decision >>>> switch to to FS. >>>> I got another challenge to throw your way. In the current fifo >>>> implementation, there's no way to identify which fifo consumer, consumes a >>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>> pass data between channels when a fifo bridge is created). I want to be able >>>> to transfer some caller information to the consumer channel on bridge, to >>>> populate an agent's screen. >>>> >>>> Under normal (non-fifo) circumstances, when a call is bridged, I've used >>>> the 'import' channel variable, so that onBridge, the aleg automatically gets >>>> populated with the bleg's 'import' field. However when fifo bridges, it does >>>> not recognize import. In other applications, I've gotten around this by >>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>> fifo::info event, but again, there's no fifo_action that directly links >>>> caller variables and consumer variables together. >>>> >>>> For now at least, I can get around this by storing uuid information in >>>> my separate database, and looking up the other channel's information based >>>> on other_leg_unique_id, but it would be nice if I could directly see it when >>>> the channel is bridged. Anyway, great program, and I hope you consider >>>> implementing these features to make FS even better. Thanks. >>>> >>>> --matt >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/08a73c72/attachment-0002.html From helmut.kuper at ewetel.de Thu Mar 26 08:23:56 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 26 Mar 2009 16:23:56 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> Message-ID: <49CB9E0C.4030300@ewetel.de> Hello Brian, On 26.03.2009 16:07, Brian West wrote: > Before we go any further... what SVN rev are you on? And by heavy > load what does your load average say? I'm using "FreeSWITCH Version 1.0.trunk (12347M)" my average load is currently this: top - 16:22:00 up 106 days, 1:49, 3 users, load average: 0.64, 0.76, 0.73 FS status is: freeswitch at internal> status UP 0 years, 3 days, 0 hours, 54 minutes, 0 seconds, 407 milliseconds, 119 microseconds 2029 session(s) since startup 4 session(s) 0/30 regards helmut From mattdfong at gmail.com Thu Mar 26 08:26:38 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 22:26:38 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> Message-ID: <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> Oh, so the reason why the bridge_api_app execution is more useful, is with the custom fifo:info event, for my event_socket to read it, it has to subscribe to ALL fifo:info events, meaning I have to process fifo:info events even if they are not useful to me. With an app in lua, I can fire a custom event based on say my fifo name, this way my event_socket only has to read events for a specific fifo, rather than all fifos. it's not to make more work for u :)...although it's sort of amazing how efficient of a coder you are. --matt On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: > Thanks of course! > But, is there any chance of firing an app? Firing an app on bridge gives > the programmer more control, rather than just listening for fifo::info > custom events. I find that lua running as a FS app can update my database > like 10x faster than reading event_socket thru Rails/Telegraph...plus, I > trust your coding much more than that of your Rail's development > counterparts. :) > > with the custom event you are firing, you should be sure to import the > variables first, then fire the event :) > > You rock Mr. Minessale > > --matt > > 2009/3/26 Anthony Minessale > > I'll fire 2 custom events when the call is bridged one for the consumer and >> one for the caller >> >> events plain custom fifo::info >> >> pull out FIFO-Name header and find the desired fifo >> pull out FIFO-Action header and look for bridge-consumer or bridge-caller >> depending on what you want to see data from. >> >> in latest trunk >> >> 2009/3/26 Matthew Fong >> >> Hi Anthony, >>> So it's been 2 days since my last request, so I'm due for another one ;) >>> >>> It would be nice if there was a way to execute a script (lua) on fifo >>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>> as my system scales, it would be better to fire a custom event. In non-fifo >>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>> fires on fifo out or fifo in channel, anything is better than having to >>> listen for a specific channel_bridge on a high-use FS installation. >>> >>> Is there anyway to get an api/script to fire on fifo bridge currently >>> that I'm missing? Thanks! >>> >>> --matt >>> >>> 2009/3/23 Anthony Minessale >>> >>> ok, >>>> maybe after this i can have a day off ;) >>>> >>>> 2 variables added to latest trunk: >>>> >>>> "fifo_caller_consumer_import" >>>> "fifo_consumer_caller_import" >>>> >>>> both work like the regular import but one is a list of vars to copy from >>>> caller to consumer and one is a list to copy from consumer to caller. >>>> >>>> >>>> 2009/3/23 Matthew Fong >>>> >>>>> Thanks Anthony, for creating the transfer_after_bridge feature for >>>>> me. Your rapid actions, are the reason I'm positive I made the right >>>>> decision switch to to FS. >>>>> I got another challenge to throw your way. In the current fifo >>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>> to transfer some caller information to the consumer channel on bridge, to >>>>> populate an agent's screen. >>>>> >>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>> does not recognize import. In other applications, I've gotten around this by >>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>> caller variables and consumer variables together. >>>>> >>>>> For now at least, I can get around this by storing uuid information in >>>>> my separate database, and looking up the other channel's information based >>>>> on other_leg_unique_id, but it would be nice if I could directly see it when >>>>> the channel is bridged. Anyway, great program, and I hope you consider >>>>> implementing these features to make FS even better. Thanks. >>>>> >>>>> --matt >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/62aabebb/attachment-0002.html From mike at jerris.com Thu Mar 26 08:28:05 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Mar 2009 11:28:05 -0400 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> Message-ID: <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> You can run a lua script (at startup or manually) that creates an event consumer to do exactly what you want. Mike On Mar 26, 2009, at 11:20 AM, Matthew Fong wrote: > Thanks of course! > > But, is there any chance of firing an app? Firing an app on bridge > gives the programmer more control, rather than just listening for > fifo::info custom events. I find that lua running as a FS app can > update my database like 10x faster than reading event_socket thru > Rails/Telegraph...plus, I trust your coding much more than that of > your Rail's development counterparts. :) > > with the custom event you are firing, you should be sure to import > the variables first, then fire the event :) > > You rock Mr. Minessale > > --matt > > 2009/3/26 Anthony Minessale > I'll fire 2 custom events when the call is bridged one for the > consumer and one for the caller > > events plain custom fifo::info > > pull out FIFO-Name header and find the desired fifo > pull out FIFO-Action header and look for bridge-consumer or bridge- > caller depending on what you want to see data from. > > in latest trunk > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/280471ac/attachment-0002.html From matt at hellohunter.com Thu Mar 26 08:30:24 2009 From: matt at hellohunter.com (Matt Hunter) Date: Thu, 26 Mar 2009 22:30:24 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> Message-ID: <4256bf830903260830y5741cad8he2374ee5e607942b@mail.gmail.com> Ahhh....can you point me to a doc or wiki, I can experiment with? --matt 2009/3/26 Michael Jerris > You can run a lua script (at startup or manually) that creates an event > consumer to do exactly what you want. > Mike > > On Mar 26, 2009, at 11:20 AM, Matthew Fong wrote: > > Thanks of course! > But, is there any chance of firing an app? Firing an app on bridge gives > the programmer more control, rather than just listening for fifo::info > custom events. I find that lua running as a FS app can update my database > like 10x faster than reading event_socket thru Rails/Telegraph...plus, I > trust your coding much more than that of your Rail's development > counterparts. :) > > with the custom event you are firing, you should be sure to import the > variables first, then fire the event :) > > You rock Mr. Minessale > > --matt > > 2009/3/26 Anthony Minessale > >> I'll fire 2 custom events when the call is bridged one for the consumer >> and one for the caller >> >> events plain custom fifo::info >> >> pull out FIFO-Name header and find the desired fifo >> pull out FIFO-Action header and look for bridge-consumer or bridge-caller >> depending on what you want to see data from. >> >> in latest trunk >> >> >> > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/5a227f84/attachment-0002.html From anthony.minessale at gmail.com Thu Mar 26 08:38:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 10:38:54 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> Message-ID: <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> this feature is already implemented system-wide not just in fifo bridge_pre_execute_aleg_app bridge_pre_execute_aleg_data bridge_pre_execute_bleg_app bridge_pre_execute_bleg_data Set either pair of these vars (aleg is the consumer) and the application of choice would be executed right when the bridge starts. 2009/3/26 Matthew Fong > Oh, so the reason why the bridge_api_app execution is more useful, is with > the custom fifo:info event, for my event_socket to read it, it has to > subscribe to ALL fifo:info events, meaning I have to process fifo:info > events even if they are not useful to me. With an app in lua, I can fire a > custom event based on say my fifo name, this way my event_socket only has to > read events for a specific fifo, rather than all fifos. > it's not to make more work for u :)...although it's sort of amazing > how efficient of a coder you are. > > --matt > > > On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: > >> Thanks of course! >> But, is there any chance of firing an app? Firing an app on bridge gives >> the programmer more control, rather than just listening for fifo::info >> custom events. I find that lua running as a FS app can update my database >> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >> trust your coding much more than that of your Rail's development >> counterparts. :) >> >> with the custom event you are firing, you should be sure to import the >> variables first, then fire the event :) >> >> You rock Mr. Minessale >> >> --matt >> >> 2009/3/26 Anthony Minessale >> >> I'll fire 2 custom events when the call is bridged one for the consumer >>> and one for the caller >>> >>> events plain custom fifo::info >>> >>> pull out FIFO-Name header and find the desired fifo >>> pull out FIFO-Action header and look for bridge-consumer or bridge-caller >>> depending on what you want to see data from. >>> >>> in latest trunk >>> >>> 2009/3/26 Matthew Fong >>> >>> Hi Anthony, >>>> So it's been 2 days since my last request, so I'm due for another one ;) >>>> >>>> It would be nice if there was a way to execute a script (lua) on fifo >>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>> fires on fifo out or fifo in channel, anything is better than having to >>>> listen for a specific channel_bridge on a high-use FS installation. >>>> >>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>> that I'm missing? Thanks! >>>> >>>> --matt >>>> >>>> 2009/3/23 Anthony Minessale >>>> >>>> ok, >>>>> maybe after this i can have a day off ;) >>>>> >>>>> 2 variables added to latest trunk: >>>>> >>>>> "fifo_caller_consumer_import" >>>>> "fifo_consumer_caller_import" >>>>> >>>>> both work like the regular import but one is a list of vars to copy >>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>> >>>>> >>>>> 2009/3/23 Matthew Fong >>>>> >>>>>> Thanks Anthony, for creating the transfer_after_bridge feature for >>>>>> me. Your rapid actions, are the reason I'm positive I made the right >>>>>> decision switch to to FS. >>>>>> I got another challenge to throw your way. In the current fifo >>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>> populate an agent's screen. >>>>>> >>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>> caller variables and consumer variables together. >>>>>> >>>>>> For now at least, I can get around this by storing uuid information in >>>>>> my separate database, and looking up the other channel's information based >>>>>> on other_leg_unique_id, but it would be nice if I could directly see it when >>>>>> the channel is bridged. Anyway, great program, and I hope you consider >>>>>> implementing these features to make FS even better. Thanks. >>>>>> >>>>>> --matt >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/22b574ae/attachment-0002.html From diego.viola at gmail.com Thu Mar 26 08:46:39 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 26 Mar 2009 11:46:39 -0400 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> Message-ID: <86a32abc0903260846t3f2390f0sa7dcb8c0230aa4df@mail.gmail.com> Use what Brian said or ssh into some Linux box and use irssi. Diego 2009/3/26 Brian West > http://cgiirc.freeswitch.org/ > I'm assume the web isn't blocked? > > /b > > On Mar 26, 2009, at 6:24 AM, Richard Lamkin wrote: > > Dear All, > > As a developer within a commercial organisation I would like to highlight > that IRC access is blocked by my organisation. This is because it falls > under the chat room category and is regarded as a security risk. > > Therefore is there any means of putting a digest of IRC traffic though the > IRC node used for Freeswitch. I like many in the commercial world are > barred by IT departmental polices from any chat room access. I feel I?m > missing out on this useful stream on information. Another issue with any > medium which is transitory is that I work in the UK and an therefore would > not be privy to communications that occur outside my time zone. > > I do support efforts to put together a forum, which although less response > than IRC is more permanent. > I am subscribed to the User and Dev mail lists which I find are a very > useful read. > > Regards > > Richard Lamkin > Mettoni Group, UK > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/eda63ebf/attachment-0002.html From matt at hellohunter.com Thu Mar 26 08:52:50 2009 From: matt at hellohunter.com (Matt Hunter) Date: Thu, 26 Mar 2009 22:52:50 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> Message-ID: <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> Ooooo, then this is an error. I'm using FreeSWITCH Version 1.0.trunk (12701M) and setting on both the fifo in and fifo out channels, but aleg.lua/bleg.lua never get executed on fifo bridge. Do you need a trace or anything? --matt 2009/3/26 Anthony Minessale > this feature is already implemented system-wide not just in fifo > > bridge_pre_execute_aleg_app > bridge_pre_execute_aleg_data > > bridge_pre_execute_bleg_app > bridge_pre_execute_bleg_data > > Set either pair of these vars (aleg is the consumer) > > and the application of choice would be executed right when the bridge > starts. > > > > 2009/3/26 Matthew Fong > >> Oh, so the reason why the bridge_api_app execution is more useful, is with >> the custom fifo:info event, for my event_socket to read it, it has to >> subscribe to ALL fifo:info events, meaning I have to process fifo:info >> events even if they are not useful to me. With an app in lua, I can fire a >> custom event based on say my fifo name, this way my event_socket only has to >> read events for a specific fifo, rather than all fifos. >> it's not to make more work for u :)...although it's sort of amazing >> how efficient of a coder you are. >> >> --matt >> >> >> On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: >> >>> Thanks of course! >>> But, is there any chance of firing an app? Firing an app on bridge gives >>> the programmer more control, rather than just listening for fifo::info >>> custom events. I find that lua running as a FS app can update my database >>> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >>> trust your coding much more than that of your Rail's development >>> counterparts. :) >>> >>> with the custom event you are firing, you should be sure to import the >>> variables first, then fire the event :) >>> >>> You rock Mr. Minessale >>> >>> --matt >>> >>> 2009/3/26 Anthony Minessale >>> >>> I'll fire 2 custom events when the call is bridged one for the consumer >>>> and one for the caller >>>> >>>> events plain custom fifo::info >>>> >>>> pull out FIFO-Name header and find the desired fifo >>>> pull out FIFO-Action header and look for bridge-consumer or >>>> bridge-caller depending on what you want to see data from. >>>> >>>> in latest trunk >>>> >>>> 2009/3/26 Matthew Fong >>>> >>>> Hi Anthony, >>>>> So it's been 2 days since my last request, so I'm due for another one >>>>> ;) >>>>> >>>>> It would be nice if there was a way to execute a script (lua) on fifo >>>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>>> fires on fifo out or fifo in channel, anything is better than having to >>>>> listen for a specific channel_bridge on a high-use FS installation. >>>>> >>>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>>> that I'm missing? Thanks! >>>>> >>>>> --matt >>>>> >>>>> 2009/3/23 Anthony Minessale >>>>> >>>>> ok, >>>>>> maybe after this i can have a day off ;) >>>>>> >>>>>> 2 variables added to latest trunk: >>>>>> >>>>>> "fifo_caller_consumer_import" >>>>>> "fifo_consumer_caller_import" >>>>>> >>>>>> both work like the regular import but one is a list of vars to copy >>>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>>> >>>>>> >>>>>> 2009/3/23 Matthew Fong >>>>>> >>>>>>> Thanks Anthony, for creating the transfer_after_bridge feature for >>>>>>> me. Your rapid actions, are the reason I'm positive I made the right >>>>>>> decision switch to to FS. >>>>>>> I got another challenge to throw your way. In the current fifo >>>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>>> populate an agent's screen. >>>>>>> >>>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>>> caller variables and consumer variables together. >>>>>>> >>>>>>> For now at least, I can get around this by storing uuid information >>>>>>> in my separate database, and looking up the other channel's information >>>>>>> based on other_leg_unique_id, but it would be nice if I could directly see >>>>>>> it when the channel is bridged. Anyway, great program, and I hope you >>>>>>> consider implementing these features to make FS even better. Thanks. >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> 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:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/5d529dd7/attachment-0002.html From mattdfong at gmail.com Thu Mar 26 08:53:49 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 22:53:49 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> Message-ID: <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> Woops, my double identity of my marketing alias isn't subscribed correctly...------------- Ooooo, then this is an error because bridge_pre_execute_aleg is not firing on fifo bridge. I'm using FreeSWITCH Version 1.0.trunk (12701M) and setting on both the fifo in and fifo out channels, but aleg.lua/bleg.lua never get executed on fifo bridge. Do you need a trace or anything? --matt On Thu, Mar 26, 2009 at 10:52 PM, Matt Hunter wrote: > > > 2009/3/26 Anthony Minessale > >> this feature is already implemented system-wide not just in fifo >> >> bridge_pre_execute_aleg_app >> bridge_pre_execute_aleg_data >> >> bridge_pre_execute_bleg_app >> bridge_pre_execute_bleg_data >> >> Set either pair of these vars (aleg is the consumer) >> >> and the application of choice would be executed right when the bridge >> starts. >> >> >> >> 2009/3/26 Matthew Fong >> >>> Oh, so the reason why the bridge_api_app execution is more useful, is >>> with the custom fifo:info event, for my event_socket to read it, it has to >>> subscribe to ALL fifo:info events, meaning I have to process fifo:info >>> events even if they are not useful to me. With an app in lua, I can fire a >>> custom event based on say my fifo name, this way my event_socket only has to >>> read events for a specific fifo, rather than all fifos. >>> it's not to make more work for u :)...although it's sort of amazing >>> how efficient of a coder you are. >>> >>> --matt >>> >>> >>> On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: >>> >>>> Thanks of course! >>>> But, is there any chance of firing an app? Firing an app on bridge gives >>>> the programmer more control, rather than just listening for fifo::info >>>> custom events. I find that lua running as a FS app can update my database >>>> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >>>> trust your coding much more than that of your Rail's development >>>> counterparts. :) >>>> >>>> with the custom event you are firing, you should be sure to import the >>>> variables first, then fire the event :) >>>> >>>> You rock Mr. Minessale >>>> >>>> --matt >>>> >>>> 2009/3/26 Anthony Minessale >>>> >>>> I'll fire 2 custom events when the call is bridged one for the consumer >>>>> and one for the caller >>>>> >>>>> events plain custom fifo::info >>>>> >>>>> pull out FIFO-Name header and find the desired fifo >>>>> pull out FIFO-Action header and look for bridge-consumer or >>>>> bridge-caller depending on what you want to see data from. >>>>> >>>>> in latest trunk >>>>> >>>>> 2009/3/26 Matthew Fong >>>>> >>>>> Hi Anthony, >>>>>> So it's been 2 days since my last request, so I'm due for another one >>>>>> ;) >>>>>> >>>>>> It would be nice if there was a way to execute a script (lua) on fifo >>>>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>>>> fires on fifo out or fifo in channel, anything is better than having to >>>>>> listen for a specific channel_bridge on a high-use FS installation. >>>>>> >>>>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>>>> that I'm missing? Thanks! >>>>>> >>>>>> --matt >>>>>> >>>>>> 2009/3/23 Anthony Minessale >>>>>> >>>>>> ok, >>>>>>> maybe after this i can have a day off ;) >>>>>>> >>>>>>> 2 variables added to latest trunk: >>>>>>> >>>>>>> "fifo_caller_consumer_import" >>>>>>> "fifo_consumer_caller_import" >>>>>>> >>>>>>> both work like the regular import but one is a list of vars to copy >>>>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>>>> >>>>>>> >>>>>>> 2009/3/23 Matthew Fong >>>>>>> >>>>>>>> Thanks Anthony, for creating the transfer_after_bridge feature for >>>>>>>> me. Your rapid actions, are the reason I'm positive I made the right >>>>>>>> decision switch to to FS. >>>>>>>> I got another challenge to throw your way. In the current fifo >>>>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>>>> populate an agent's screen. >>>>>>>> >>>>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>>>> caller variables and consumer variables together. >>>>>>>> >>>>>>>> For now at least, I can get around this by storing uuid information >>>>>>>> in my separate database, and looking up the other channel's information >>>>>>>> based on other_leg_unique_id, but it would be nice if I could directly see >>>>>>>> it when the channel is bridged. Anyway, great program, and I hope you >>>>>>>> consider implementing these features to make FS even better. Thanks. >>>>>>>> >>>>>>>> --matt >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> >>>>>>> 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:213-799-1400 >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> 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:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/38f794a2/attachment-0002.html From brian at freeswitch.org Thu Mar 26 08:33:43 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 10:33:43 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260830y5741cad8he2374ee5e607942b@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> <4256bf830903260830y5741cad8he2374ee5e607942b@mail.gmail.com> Message-ID: <9A65047C-F6E8-4298-A4B9-66193159C619@freeswitch.org> http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.EventConsumer On Mar 26, 2009, at 10:30 AM, Matt Hunter wrote: > Ahhh....can you point me to a doc or wiki, I can experiment with? > From mattdfong at gmail.com Thu Mar 26 09:00:23 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 23:00:23 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <9A65047C-F6E8-4298-A4B9-66193159C619@freeswitch.org> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <62BECEF1-3A33-4C00-837F-8AD5D6F35836@jerris.com> <4256bf830903260830y5741cad8he2374ee5e607942b@mail.gmail.com> <9A65047C-F6E8-4298-A4B9-66193159C619@freeswitch.org> Message-ID: <4256bf830903260900l7d152800p9678663b4abe28ed@mail.gmail.com> Hi Brian, Thanks for the link...I saw that, but i'm a newbie to lua (only use it cause of FS), and I'm a little confused how the example works. It consumes all events? Then subscribes to a session? and then, every second checks to see if an event has been fired for that session? Would it be possible to get an idea of how to subscribe to all events, and have a function execute for each time an event is fired? Can lua "wait" until an event is fired, or must it loop and sleep every second? Thanks for the help. --matt --- the example... con = freeswitch.EventConsumer("all"); session = freeswitch.Session("sofia/default/dest at host.com"); while session:ready() do session:execute("sleep", "1000"); for e in (function() return con:pop() end) do print("event\n" .. e:serialize("xml")); end end 2009/3/26 Brian West > http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.EventConsumer > > On Mar 26, 2009, at 10:30 AM, Matt Hunter wrote: > > > Ahhh....can you point me to a doc or wiki, I can experiment with? > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/b96eb8c5/attachment-0002.html From timr at asteriasgi.com Thu Mar 26 09:01:15 2009 From: timr at asteriasgi.com (Tim Ringenbach) Date: Thu, 26 Mar 2009 11:01:15 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> Message-ID: <49CBA6CB.4030005@asteriasgi.com> Is there nothing out there that integrates a forum with a mailing list? It seems like one could display the mailing list archives exactly like a forum, and allow users to register to the forum and post (appearing to the mailing list as username at forumurl.org) in such a way that they don't have to realize it's a mailing list. Anthony Minessale wrote: > The guy started a forum almost a month ago and as you can see nobody > knows the url and it has no posts. > > http://freeswitch411.info/forum/ > > This is one of the problems I was worried about when endorsing a forum. > From anthony.minessale at gmail.com Thu Mar 26 09:04:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 11:04:11 -0500 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> Message-ID: <191c3a030903260904m466e10c0r3df8c38ee690b4ad@mail.gmail.com> see below 2009/3/26 Matthew Fong > Woops, my double identity of my marketing alias isn't subscribed > correctly...------------- > > Ooooo, then this is an error because bridge_pre_execute_aleg is not firing > on fifo bridge. I'm using > FreeSWITCH Version 1.0.trunk (12701M) > > and setting > > > data="bridge_pre_execute_aleg_app=aleg.lua"/> <-- aleg_data not aleg_app > again...... > > data="bridge_pre_execute_bleg_app=bleg.lua"/> <-- bleg_data not bleg_app > again > > on both the fifo in and fifo out channels, but aleg.lua/bleg.lua never get > executed on fifo bridge. Do you need a trace or anything? > > --matt > On Thu, Mar 26, 2009 at 10:52 PM, Matt Hunter wrote: > >> >> >> 2009/3/26 Anthony Minessale >> >>> this feature is already implemented system-wide not just in fifo >>> >>> bridge_pre_execute_aleg_app >>> bridge_pre_execute_aleg_data >>> >>> bridge_pre_execute_bleg_app >>> bridge_pre_execute_bleg_data >>> >>> Set either pair of these vars (aleg is the consumer) >>> >>> and the application of choice would be executed right when the bridge >>> starts. >>> >>> >>> >>> 2009/3/26 Matthew Fong >>> >>>> Oh, so the reason why the bridge_api_app execution is more useful, is >>>> with the custom fifo:info event, for my event_socket to read it, it has to >>>> subscribe to ALL fifo:info events, meaning I have to process fifo:info >>>> events even if they are not useful to me. With an app in lua, I can fire a >>>> custom event based on say my fifo name, this way my event_socket only has to >>>> read events for a specific fifo, rather than all fifos. >>>> it's not to make more work for u :)...although it's sort of amazing >>>> how efficient of a coder you are. >>>> >>>> --matt >>>> >>>> >>>> On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: >>>> >>>>> Thanks of course! >>>>> But, is there any chance of firing an app? Firing an app on bridge >>>>> gives the programmer more control, rather than just listening for fifo::info >>>>> custom events. I find that lua running as a FS app can update my database >>>>> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >>>>> trust your coding much more than that of your Rail's development >>>>> counterparts. :) >>>>> >>>>> with the custom event you are firing, you should be sure to import the >>>>> variables first, then fire the event :) >>>>> >>>>> You rock Mr. Minessale >>>>> >>>>> --matt >>>>> >>>>> 2009/3/26 Anthony Minessale >>>>> >>>>> I'll fire 2 custom events when the call is bridged one for the consumer >>>>>> and one for the caller >>>>>> >>>>>> events plain custom fifo::info >>>>>> >>>>>> pull out FIFO-Name header and find the desired fifo >>>>>> pull out FIFO-Action header and look for bridge-consumer or >>>>>> bridge-caller depending on what you want to see data from. >>>>>> >>>>>> in latest trunk >>>>>> >>>>>> 2009/3/26 Matthew Fong >>>>>> >>>>>> Hi Anthony, >>>>>>> So it's been 2 days since my last request, so I'm due for another one >>>>>>> ;) >>>>>>> >>>>>>> It would be nice if there was a way to execute a script (lua) on fifo >>>>>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>>>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>>>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>>>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>>>>> fires on fifo out or fifo in channel, anything is better than having to >>>>>>> listen for a specific channel_bridge on a high-use FS installation. >>>>>>> >>>>>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>>>>> that I'm missing? Thanks! >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> 2009/3/23 Anthony Minessale >>>>>>> >>>>>>> ok, >>>>>>>> maybe after this i can have a day off ;) >>>>>>>> >>>>>>>> 2 variables added to latest trunk: >>>>>>>> >>>>>>>> "fifo_caller_consumer_import" >>>>>>>> "fifo_consumer_caller_import" >>>>>>>> >>>>>>>> both work like the regular import but one is a list of vars to copy >>>>>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>>>>> >>>>>>>> >>>>>>>> 2009/3/23 Matthew Fong >>>>>>>> >>>>>>>>> Thanks Anthony, for creating the transfer_after_bridge feature >>>>>>>>> for me. Your rapid actions, are the reason I'm positive I made the right >>>>>>>>> decision switch to to FS. >>>>>>>>> I got another challenge to throw your way. In the current fifo >>>>>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>>>>> populate an agent's screen. >>>>>>>>> >>>>>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>>>>> caller variables and consumer variables together. >>>>>>>>> >>>>>>>>> For now at least, I can get around this by storing uuid information >>>>>>>>> in my separate database, and looking up the other channel's information >>>>>>>>> based on other_leg_unique_id, but it would be nice if I could directly see >>>>>>>>> it when the channel is bridged. Anyway, great program, and I hope you >>>>>>>>> consider implementing these features to make FS even better. Thanks. >>>>>>>>> >>>>>>>>> --matt >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II >>>>>>>> >>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>> >>>>>>>> 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:213-799-1400 >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/cd092047/attachment-0002.html From Richard.Lamkin at mettoni.com Thu Mar 26 09:05:16 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 26 Mar 2009 16:05:16 -0000 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com><3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <5D5871D6-89FD-44DE-9C1A-AD0C7916DBFA@freeswitch.org> Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804B2AF06@nickel.mettonigroup.com> Brian, Thank you this worked for me; I have put these details in the Freeswitch wiki IRC page Regards Richard Lamkin From: Brian West [mailto:brian at freeswitch.org] Sent: 26 March 2009 14:33 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] IRC is not for all http://cgiirc.freeswitch.org/ I'm assume the web isn't blocked? /b On Mar 26, 2009, at 6:24 AM, Richard Lamkin wrote: Dear All, As a developer within a commercial organisation I would like to highlight that IRC access is blocked by my organisation. This is because it falls under the chat room category and is regarded as a security risk. Therefore is there any means of putting a digest of IRC traffic though the IRC node used for Freeswitch. I like many in the commercial world are barred by IT departmental polices from any chat room access. I feel I'm missing out on this useful stream on information. Another issue with any medium which is transitory is that I work in the UK and an therefore would not be privy to communications that occur outside my time zone. I do support efforts to put together a forum, which although less response than IRC is more permanent. I am subscribed to the User and Dev mail lists which I find are a very useful read. Regards Richard Lamkin Mettoni Group, UK ************************************************************************ * 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************ * _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/2fe3d4a4/attachment-0002.html From anthony.minessale at gmail.com Thu Mar 26 09:05:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 11:05:19 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <49CBA6CB.4030005@asteriasgi.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> Message-ID: <191c3a030903260905y389dadc0mb1c55636ce87dbf9@mail.gmail.com> That is our plan (nabble does this) On Thu, Mar 26, 2009 at 11:01 AM, Tim Ringenbach wrote: > Is there nothing out there that integrates a forum with a mailing list? > It seems like one could display the mailing list archives exactly like a > forum, and allow users to register to the forum and post (appearing to > the mailing list as username at forumurl.org) in such a way that they don't > have to realize it's a mailing list. > > Anthony Minessale wrote: > > The guy started a forum almost a month ago and as you can see nobody > > knows the url and it has no posts. > > > > http://freeswitch411.info/forum/ > > > > This is one of the problems I was worried about when endorsing a forum. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/9acb255e/attachment-0002.html From mike at jerris.com Thu Mar 26 09:06:54 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Mar 2009 12:06:54 -0400 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <49CBA6CB.4030005@asteriasgi.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> Message-ID: http://n2.nabble.com/freeswitch-users-f2379917.html Mike On Mar 26, 2009, at 12:01 PM, Tim Ringenbach wrote: > Is there nothing out there that integrates a forum with a mailing > list? > It seems like one could display the mailing list archives exactly > like a > forum, and allow users to register to the forum and post (appearing to > the mailing list as username at forumurl.org) in such a way that they > don't > have to realize it's a mailing list. > > Anthony Minessale wrote: >> The guy started a forum almost a month ago and as you can see nobody >> knows the url and it has no posts. >> >> http://freeswitch411.info/forum/ >> >> This is one of the problems I was worried about when endorsing a >> forum. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Thu Mar 26 09:07:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Mar 2009 09:07:59 -0700 Subject: [Freeswitch-users] Action and Anti-Action In-Reply-To: <49CB90D5.4090901@devel-it.com.br> References: <49CB90D5.4090901@devel-it.com.br> Message-ID: <87f2f3b90903260907i3568f8edt9794ad852c514656@mail.gmail.com> Look in the default.xml dialplan file for the "tod_example" extension. (It's near the top of the file.) It has an example of how to create an extension that simply sets a variable that can be used in other extensions in the dialplan. -MC 2009/3/26 Rodrigo P. Telles : > Hi Guys, > > I'm trying to do some string matching against a created var and looks like I > am doing something wrong but I can't find whats it. > > I'm wrote an extension just for tests purposes on dialplan/default.xml: > > > ???? > ??????? > ???? > ???? > ?????? > ?????? > ?????? > ?????? > ???? > > > Using two SIP extensions (1000 and 1001) behind NAT and I expected too see > "Action=1" on the logs/console, but I'm seeing "Anti-Action=1". > > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x Execute > set(nated=${cond(${network_addr} != ${sip_contact_host} ? 1 : 0)}) > 2009-03-26 11:02:57 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1001 at x.x.x.x Expanded String > set(nated=1) > 2009-03-26 11:02:57 [DEBUG] mod_dptools.c:711 set_function() > sofia/internal/1001 at x.x.x.x SET [nated]=[1] > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x Execute > log(Anti-Action=${nated}) > 2009-03-26 11:02:57 [DEBUG] switch_core_session.c:1286 > switch_core_session_exec() sofia/internal/1001 at x.x.x.x Expanded String > log(Anti-Action=1) > 2009-03-26 11:02:57 [DEBUG] mod_dptools.c:895 log_function() Anti-Action=1 > 2009-03-26 11:02:57 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/1001 at x.x.x.x Execute > set(dialed_extension=1000) > .... > > I really appreciate any inputs. > I'm using FS 1.0.3 stable. > > regards, > Rodrigo Telles > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Mar 26 09:13:54 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 11:13:54 -0500 Subject: [Freeswitch-users] Action and Anti-Action In-Reply-To: <87f2f3b90903260907i3568f8edt9794ad852c514656@mail.gmail.com> References: <49CB90D5.4090901@devel-it.com.br> <87f2f3b90903260907i3568f8edt9794ad852c514656@mail.gmail.com> Message-ID: Remember the dialplan is parsed, installed into the session then sent into the execute state. The dialplan is NOT execute line by line as it goes thru the dialplan. Which means you can not set a var on one line then condition on that var on the next line because the var isn't set yet... Remember its like a todo list one by one its compiled but not executed yet. This is usually an alien concept when you're migrating from various other telephony software. /b On Mar 26, 2009, at 11:07 AM, Michael Collins wrote: > Look in the default.xml dialplan file for the "tod_example" extension. > (It's near the top of the file.) It has an example of how to create an > extension that simply sets a variable that can be used in other > extensions in the dialplan. > > -MC Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/f4c195f1/attachment-0002.html From mattdfong at gmail.com Thu Mar 26 09:15:59 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Thu, 26 Mar 2009 23:15:59 +0700 Subject: [Freeswitch-users] Another fifo request In-Reply-To: <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> References: <4256bf830903230849p5cd3d326rea0f38803c99bd2f@mail.gmail.com> <191c3a030903230908p7d87fbb9s53bbc337c1c022f3@mail.gmail.com> <4256bf830903260240n34ca9b27u9e5fafcaa4dda6b2@mail.gmail.com> <191c3a030903260807v705ddbd9o5053249a2e771805@mail.gmail.com> <4256bf830903260820r4bee5bd7ga8df4a6148627b7e@mail.gmail.com> <4256bf830903260826h1dcb24eev68234785b791fe18@mail.gmail.com> <191c3a030903260838i423222ffv8f3dd6a176a5e437@mail.gmail.com> <4256bf830903260852gfdde108m3b9922eba18d869d@mail.gmail.com> <4256bf830903260853j288b1133y70fc309cf660b18@mail.gmail.com> Message-ID: <4256bf830903260915g6c47f256k55aab73ddc8a9f25@mail.gmail.com> Yah, your right. it works...but it must be set on the fifo out channel (consumer's channel), it will not execute if it's set on the fifo in channel (caller's channel). Also api_after_bridge does not execute...but as long as bridge_pre_execute_a/bleg works, I'm super happy. Thanks. --matt On Thu, Mar 26, 2009 at 10:53 PM, Matthew Fong wrote: > Woops, my double identity of my marketing alias isn't subscribed > correctly...------------- > > Ooooo, then this is an error because bridge_pre_execute_aleg is not firing > on fifo bridge. I'm using > FreeSWITCH Version 1.0.trunk (12701M) > > and setting > > > data="bridge_pre_execute_aleg_app=aleg.lua"/> > > data="bridge_pre_execute_bleg_app=bleg.lua"/> > > on both the fifo in and fifo out channels, but aleg.lua/bleg.lua never get > executed on fifo bridge. Do you need a trace or anything? > > --matt > On Thu, Mar 26, 2009 at 10:52 PM, Matt Hunter wrote: > >> >> >> 2009/3/26 Anthony Minessale >> >>> this feature is already implemented system-wide not just in fifo >>> >>> bridge_pre_execute_aleg_app >>> bridge_pre_execute_aleg_data >>> >>> bridge_pre_execute_bleg_app >>> bridge_pre_execute_bleg_data >>> >>> Set either pair of these vars (aleg is the consumer) >>> >>> and the application of choice would be executed right when the bridge >>> starts. >>> >>> >>> >>> 2009/3/26 Matthew Fong >>> >>>> Oh, so the reason why the bridge_api_app execution is more useful, is >>>> with the custom fifo:info event, for my event_socket to read it, it has to >>>> subscribe to ALL fifo:info events, meaning I have to process fifo:info >>>> events even if they are not useful to me. With an app in lua, I can fire a >>>> custom event based on say my fifo name, this way my event_socket only has to >>>> read events for a specific fifo, rather than all fifos. >>>> it's not to make more work for u :)...although it's sort of amazing >>>> how efficient of a coder you are. >>>> >>>> --matt >>>> >>>> >>>> On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong wrote: >>>> >>>>> Thanks of course! >>>>> But, is there any chance of firing an app? Firing an app on bridge >>>>> gives the programmer more control, rather than just listening for fifo::info >>>>> custom events. I find that lua running as a FS app can update my database >>>>> like 10x faster than reading event_socket thru Rails/Telegraph...plus, I >>>>> trust your coding much more than that of your Rail's development >>>>> counterparts. :) >>>>> >>>>> with the custom event you are firing, you should be sure to import the >>>>> variables first, then fire the event :) >>>>> >>>>> You rock Mr. Minessale >>>>> >>>>> --matt >>>>> >>>>> 2009/3/26 Anthony Minessale >>>>> >>>>> I'll fire 2 custom events when the call is bridged one for the consumer >>>>>> and one for the caller >>>>>> >>>>>> events plain custom fifo::info >>>>>> >>>>>> pull out FIFO-Name header and find the desired fifo >>>>>> pull out FIFO-Action header and look for bridge-consumer or >>>>>> bridge-caller depending on what you want to see data from. >>>>>> >>>>>> in latest trunk >>>>>> >>>>>> 2009/3/26 Matthew Fong >>>>>> >>>>>> Hi Anthony, >>>>>>> So it's been 2 days since my last request, so I'm due for another one >>>>>>> ;) >>>>>>> >>>>>>> It would be nice if there was a way to execute a script (lua) on fifo >>>>>>> bridge. I currently rely on the channel_bridge event, but I'm worried that >>>>>>> as my system scales, it would be better to fire a custom event. In non-fifo >>>>>>> mode, I can do this with bridge_pre_execute_bleg_app/data, but this >>>>>>> doesn't work with a fifo bridge. It doesn't really matter which channel it >>>>>>> fires on fifo out or fifo in channel, anything is better than having to >>>>>>> listen for a specific channel_bridge on a high-use FS installation. >>>>>>> >>>>>>> Is there anyway to get an api/script to fire on fifo bridge currently >>>>>>> that I'm missing? Thanks! >>>>>>> >>>>>>> --matt >>>>>>> >>>>>>> 2009/3/23 Anthony Minessale >>>>>>> >>>>>>> ok, >>>>>>>> maybe after this i can have a day off ;) >>>>>>>> >>>>>>>> 2 variables added to latest trunk: >>>>>>>> >>>>>>>> "fifo_caller_consumer_import" >>>>>>>> "fifo_consumer_caller_import" >>>>>>>> >>>>>>>> both work like the regular import but one is a list of vars to copy >>>>>>>> from caller to consumer and one is a list to copy from consumer to caller. >>>>>>>> >>>>>>>> >>>>>>>> 2009/3/23 Matthew Fong >>>>>>>> >>>>>>>>> Thanks Anthony, for creating the transfer_after_bridge feature >>>>>>>>> for me. Your rapid actions, are the reason I'm positive I made the right >>>>>>>>> decision switch to to FS. >>>>>>>>> I got another challenge to throw your way. In the current fifo >>>>>>>>> implementation, there's no way to identify which fifo consumer, consumes a >>>>>>>>> caller--besides using other_leg_unique_id on bridge (ie, there's no way to >>>>>>>>> pass data between channels when a fifo bridge is created). I want to be able >>>>>>>>> to transfer some caller information to the consumer channel on bridge, to >>>>>>>>> populate an agent's screen. >>>>>>>>> >>>>>>>>> Under normal (non-fifo) circumstances, when a call is bridged, I've >>>>>>>>> used the 'import' channel variable, so that onBridge, the aleg automatically >>>>>>>>> gets populated with the bleg's 'import' field. However when fifo bridges, it >>>>>>>>> does not recognize import. In other applications, I've gotten around this by >>>>>>>>> using bridge_pre_execute_bleg_app/data to throw a custom event but with >>>>>>>>> fifo, bridge_pre_execute also does not work. I've been looking at the >>>>>>>>> fifo::info event, but again, there's no fifo_action that directly links >>>>>>>>> caller variables and consumer variables together. >>>>>>>>> >>>>>>>>> For now at least, I can get around this by storing uuid information >>>>>>>>> in my separate database, and looking up the other channel's information >>>>>>>>> based on other_leg_unique_id, but it would be nice if I could directly see >>>>>>>>> it when the channel is bridged. Anyway, great program, and I hope you >>>>>>>>> consider implementing these features to make FS even better. Thanks. >>>>>>>>> >>>>>>>>> --matt >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II >>>>>>>> >>>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>>> ClueCon http://www.cluecon.com/ >>>>>>>> >>>>>>>> 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:213-799-1400 >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> 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:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> 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:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/006e694a/attachment-0002.html From jonas.gauffin at gmail.com Thu Mar 26 11:09:55 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Thu, 26 Mar 2009 19:09:55 +0100 Subject: [Freeswitch-users] sipp emulating a registered end point Message-ID: Hello I want to achive this: Sipp1 -> FS -> Sipp2 Sipp1 emulates a inbound calls (easy to achive) Sipp2 should emulate a registered user (i.e. register with FS and then just wait for calls and hangup when sipp1 hangsup) How do I configure sipp as "Sipp2"? Thanks, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/12d72e84/attachment-0002.html From Richard.Lamkin at mettoni.com Thu Mar 26 11:26:42 2009 From: Richard.Lamkin at mettoni.com (Richard Lamkin) Date: Thu, 26 Mar 2009 18:26:42 -0000 Subject: [Freeswitch-users] Load testing and thread use Message-ID: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> Dear All, I am testing FS as a call queuing server. My test set up is as follows; Two Windows XP (SP3) Pc's with 1.0.3 installed out of the box. [ Before anyone says use Linux I need to use windows for a specific reason] I have a single gateway [fsb1500] on FS-1 configured to register with FS-2 [extn 1500]. I then use the following CLI command to create a call from FS1[5900] to FS2[5900] "bgapi originate sofia/gateway/fsb1500/5900 at richardl-5013-2.mettonigroup.com 5900" This happily connects FS1[5900] to FS2[5900] . Codec set to G711a. I have one hardphone on FS2 with a call connected to 5900 to allow me to listen to MOH. I have been monitoring the system resources used by FS and have observed that for each new call a new thread is created. I have pushed well over 200 calls from FS-1 into FS2. All goes well with the CPU usage by FS (<25%) until 150 concurrent calls when it starts to climb rapidly and at 175 (~50%). One good thing is that memory usage is reasonable with 90MB for 150 calls. My questions are; Q1 - Is my test reasonable ? Q2 - Windows XP is not that happy with large numbers of thread per app. Is there a way to configure FS to use fewer threads; instead of using 1/call can it be configured to use to 1/(n calls) ? Q3 - If there's not a way to configure FS to combine threads would it be reasonable to consider modifying the FS code to allow the combining of threads or would this be a mammoth task. Regards Richard Lamkin ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/bcd253b4/attachment-0002.html From solko at gcdf.pl Thu Mar 26 12:46:32 2009 From: solko at gcdf.pl (Szymon Olko) Date: Thu, 26 Mar 2009 20:46:32 +0100 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> Message-ID: <49CBDB98.4090404@gcdf.pl> Richard Lamkin pisze: > Dear All, > > > > I am testing FS as a call queuing server. > > > > My test set up is as follows; > > Two Windows XP (SP3) Pc?s with 1.0.3 installed out of the box. [ > Before anyone says use Linux I need to use windows for a specific reason] > > > > I have a single gateway [fsb1500] on FS-1 configured to register with > FS-2 [extn 1500]. > > > > I then use the following CLI command to create a call from FS1[5900] to > FS2[5900] > > ?bgapi originate > sofia/gateway/fsb1500/5900 at richardl-5013-2.mettonigroup.com 5900? > > > > This happily connects FS1[5900] to FS2[5900] . Codec set to G711a. > > I have one hardphone on FS2 with a call connected to 5900 to allow me to > listen to MOH. > > > > I have been monitoring the system resources used by FS and have observed > that for each new call a new thread is created. > > I have pushed well over 200 calls from FS-1 into FS2. > > All goes well with the CPU usage by FS (<25%) until 150 concurrent calls > when it starts to climb rapidly and at 175 (~50%). > > One good thing is that memory usage is reasonable with 90MB for 150 calls. > > > > My questions are; > > Q1 ? Is my test reasonable ? > Memory usage looks good, I noticed on my laptop with Linux with Pentium M 1,6GHz that having something like 200 channels connected makes very big CPU usage, but what to expect if every channel was connected to one channel. > > > Q2 - Windows XP is not that happy with large numbers of thread per app. > Is there a way to configure FS to use fewer threads; instead of using > 1/call can it be configured to use to 1/(n calls) ? > Anthony here describes that there is one thread for channel, so don't expect it will change. > > > Q3 ? If there?s not a way to configure FS to combine threads would it > be reasonable to consider modifying the FS code to allow the combining > of threads or would this be a mammoth task. > The same as above. I don't know XP so well, but if it only has problems with many threads for one application then consider running multiple FS instances on one machine. > > > Regards > > Szymon Olko > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Mar 26 13:20:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Mar 2009 15:20:31 -0500 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> Message-ID: <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> The core philosophy of the entire design is based on a single thread per channel. There is no way to avoid this. What are the specs of your machines? DId you try using the 2003 server instead of XP (a home version meant for single users) We have little feedback on performance on win32 (good or bad) but we do have many people using it. Did you do any more extensive profiling to see what is using the most cpu (look for process explorer) 2009/3/26 Richard Lamkin > Dear All, > > > > I am testing FS as a call queuing server. > > > > My test set up is as follows; > > Two Windows XP (SP3) Pc?s with 1.0.3 installed out of the box. [ Before > anyone says use Linux I need to use windows for a specific reason] > > > > I have a single gateway [fsb1500] on FS-1 configured to register with FS-2 > [extn 1500]. > > > > I then use the following CLI command to create a call from FS1[5900] to > FS2[5900] > > ?bgapi originate sofia/gateway/fsb1500/ > 5900 at richardl-5013-2.mettonigroup.com 5900? > > > > This happily connects FS1[5900] to FS2[5900] . Codec set to G711a. > > I have one hardphone on FS2 with a call connected to 5900 to allow me to > listen to MOH. > > > > I have been monitoring the system resources used by FS and have observed > that for each new call a new thread is created. > > I have pushed well over 200 calls from FS-1 into FS2. > > All goes well with the CPU usage by FS (<25%) until 150 concurrent calls > when it starts to climb rapidly and at 175 (~50%). > > One good thing is that memory usage is reasonable with 90MB for 150 calls. > > > > My questions are; > > Q1 ? Is my test reasonable ? > > > > Q2 - Windows XP is not that happy with large numbers of thread per app. Is > there a way to configure FS to use fewer threads; instead of using 1/call > can it be configured to use to 1/(n calls) ? > > > > Q3 ? If there?s not a way to configure FS to combine threads would it be > reasonable to consider modifying the FS code to allow the combining of > threads or would this be a mammoth task. > > > > Regards > > > > Richard Lamkin > > ************************************************************************* > 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 > > Datapulse Ltd (part of the Mettoni Group) > Registered in England and Wales: 4485978 > 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN > ************************************************************************* > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/0104b63f/attachment-0002.html From Prometheus001 at gmx.net Thu Mar 26 13:50:21 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 26 Mar 2009 21:50:21 +0100 Subject: [Freeswitch-users] Originate and Conference Message-ID: <49CBEA8D.4050901@gmx.net> Hello, when I originate a call via event socket and transfer it to a conference, it doesn't ask for a PIN. Example: originate sofia/default/222331 &conference(222500) The same happens when I originate a call and transfer it to the 222500 destination, which then transfers the call to the conference. originate sofia/default/222331 &transfer(222500) If I manually dial from the phone (222331) to the conference it correctly asks for the PIN. Anybody has a clue why this happens and how to enable the PIN while originating? Best regards Peter From msc at freeswitch.org Thu Mar 26 14:32:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Mar 2009 14:32:21 -0700 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <49CBEA8D.4050901@gmx.net> References: <49CBEA8D.4050901@gmx.net> Message-ID: <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> On Thu, Mar 26, 2009 at 1:50 PM, Peter P GMX wrote: > Hello, > > when I originate a call via event socket and transfer it to a > conference, it doesn't ask for a PIN. > Example: > originate sofia/default/222331 &conference(222500) > The same happens when I originate a call and transfer it to the 222500 > destination, which then transfers the call to the conference. > originate sofia/default/222331 &transfer(222500) > > If I manually dial from the phone (222331) to the conference it > correctly asks for the PIN. > > Anybody has a clue why this happens and how to enable the PIN while > originating? Is 222500 part of your dialplan? If so try this: originate sofia/default/222331 222500 Let us know if that works. -MC From peter at cindyandpeter.com Thu Mar 26 14:40:16 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Thu, 26 Mar 2009 17:40:16 -0400 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> Message-ID: <038c01c9ae5b$743cd310$5cb67930$@com> Anthony, No argument on the core design - this is what differentiates FS from * and what makes things solid. But, on Windows at least, the "one-thread-per-client" concept normally only scales so far (Server normally being a bit better than XP). Not sure about their availability on other platforms, but have you looked at fibers? They seem to be a lighter weight alternative with most of the benefits. Peter From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, March 26, 2009 4:21 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Load testing and thread use The core philosophy of the entire design is based on a single thread per channel. There is no way to avoid this. What are the specs of your machines? DId you try using the 2003 server instead of XP (a home version meant for single users) We have little feedback on performance on win32 (good or bad) but we do have many people using it. Did you do any more extensive profiling to see what is using the most cpu (look for process explorer) 2009/3/26 Richard Lamkin Dear All, I am testing FS as a call queuing server. My test set up is as follows; Two Windows XP (SP3) Pc's with 1.0.3 installed out of the box. [ Before anyone says use Linux I need to use windows for a specific reason] I have a single gateway [fsb1500] on FS-1 configured to register with FS-2 [extn 1500]. I then use the following CLI command to create a call from FS1[5900] to FS2[5900] "bgapi originate sofia/gateway/fsb1500/5900 at richardl-5013-2.mettonigroup.com 5900" This happily connects FS1[5900] to FS2[5900] . Codec set to G711a. I have one hardphone on FS2 with a call connected to 5900 to allow me to listen to MOH. I have been monitoring the system resources used by FS and have observed that for each new call a new thread is created. I have pushed well over 200 calls from FS-1 into FS2. All goes well with the CPU usage by FS (<25%) until 150 concurrent calls when it starts to climb rapidly and at 175 (~50%). One good thing is that memory usage is reasonable with 90MB for 150 calls. My questions are; Q1 - Is my test reasonable ? Q2 - Windows XP is not that happy with large numbers of thread per app. Is there a way to configure FS to use fewer threads; instead of using 1/call can it be configured to use to 1/(n calls) ? Q3 - If there's not a way to configure FS to combine threads would it be reasonable to consider modifying the FS code to allow the combining of threads or would this be a mammoth task. Regards Richard Lamkin ************************************************************************* 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 Datapulse Ltd (part of the Mettoni Group) Registered in England and Wales: 4485978 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/879aa827/attachment-0002.html From brian at freeswitch.org Thu Mar 26 15:02:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 17:02:16 -0500 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <038c01c9ae5b$743cd310$5cb67930$@com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> <038c01c9ae5b$743cd310$5cb67930$@com> Message-ID: <526F1BEB-EC70-4234-976F-7F8777A89065@freeswitch.org> Is this windows specific? /b On Mar 26, 2009, at 4:40 PM, Peter J. Zandvoort wrote: > have you looked at fibers? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/b62ce3ef/attachment-0002.html From brian at freeswitch.org Thu Mar 26 15:05:27 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Mar 2009 17:05:27 -0500 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <038c01c9ae5b$743cd310$5cb67930$@com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> <038c01c9ae5b$743cd310$5cb67930$@com> Message-ID: <7EFF1BA2-BA80-4454-9E77-A90F4452E987@freeswitch.org> OH so this is like coroutines. /b On Mar 26, 2009, at 4:40 PM, Peter J. Zandvoort wrote: > have you looked at fibers? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/73d3212b/attachment-0002.html From grevenx at me.com Thu Mar 26 15:12:08 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Thu, 26 Mar 2009 23:12:08 +0100 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <7EFF1BA2-BA80-4454-9E77-A90F4452E987@freeswitch.org> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> <038c01c9ae5b$743cd310$5cb67930$@com> <7EFF1BA2-BA80-4454-9E77-A90F4452E987@freeswitch.org> Message-ID: <6E8BEF52-E107-45E9-84D7-D3F182203C2B@me.com> http://en.wikipedia.org/wiki/Fiber_(computer_science) Even Andr? Fiskvik On 26. mars. 2009, at 23.05, Brian West wrote: > OH so this is like coroutines. > > /b > > On Mar 26, 2009, at 4:40 PM, Peter J. Zandvoort wrote: > >> have you looked at fibers? > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/c64318dd/attachment-0002.html From telles-listas at devel-it.com.br Thu Mar 26 15:25:17 2009 From: telles-listas at devel-it.com.br (Rodrigo P. Telles) Date: Thu, 26 Mar 2009 19:25:17 -0300 Subject: [Freeswitch-users] Action and Anti-Action In-Reply-To: References: <49CB90D5.4090901@devel-it.com.br> Message-ID: <49CC00CD.6090703@devel-it.com.br> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/6c6ccdff/attachment-0002.html From peter at cindyandpeter.com Thu Mar 26 15:33:29 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Thu, 26 Mar 2009 18:33:29 -0400 Subject: [Freeswitch-users] Load testing and thread use In-Reply-To: <6E8BEF52-E107-45E9-84D7-D3F182203C2B@me.com> References: <3181A30B8C35AB4AA8577B78DDF4613804B2B00C@nickel.mettonigroup.com> <191c3a030903261320x525a6ebdt80ddeb8b068cb78@mail.gmail.com> <038c01c9ae5b$743cd310$5cb67930$@com> <7EFF1BA2-BA80-4454-9E77-A90F4452E987@freeswitch.org> <6E8BEF52-E107-45E9-84D7-D3F182203C2B@me.com> Message-ID: <03ad01c9ae62$e37a2230$aa6e6690$@com> Looks like it! I must admit that I have very little experience with it . it just looked like an additional toy to add to the bag of tricks. I?ve seen it used on mail servers and I think SQL Server now has the option. Both sound like things that would have long running client connections for which you want the simplicity of dedicated threads but not the overhead. For Windows: http://msdn.microsoft.com/en-us/library/ms682115(VS.85).aspx From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Even Andr? Fiskvik Sent: Thursday, March 26, 2009 6:12 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Load testing and thread use http://en.wikipedia.org/wiki/Fiber_(computer_science) Even Andr? Fiskvik On 26. mars. 2009, at 23.05, Brian West wrote: OH so this is like coroutines. /b On Mar 26, 2009, at 4:40 PM, Peter J. Zandvoort wrote: have you looked at fibers? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090326/bc00b04a/attachment-0002.html From Prometheus001 at gmx.net Thu Mar 26 16:09:59 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Mar 2009 00:09:59 +0100 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> Message-ID: <49CC0B47.6000508@gmx.net> Hello Michael, I tried this, but received the same behaviour. It does not ask for the defined PIN. Best regards Peter Michael Collins schrieb: > On Thu, Mar 26, 2009 at 1:50 PM, Peter P GMX wrote: > >> Hello, >> >> when I originate a call via event socket and transfer it to a >> conference, it doesn't ask for a PIN. >> Example: >> originate sofia/default/222331 &conference(222500) >> The same happens when I originate a call and transfer it to the 222500 >> destination, which then transfers the call to the conference. >> originate sofia/default/222331 &transfer(222500) >> >> If I manually dial from the phone (222331) to the conference it >> correctly asks for the PIN. >> >> Anybody has a clue why this happens and how to enable the PIN while >> originating? >> > > Is 222500 part of your dialplan? If so try this: > > originate sofia/default/222331 222500 > > Let us know if that works. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu Mar 26 16:58:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Mar 2009 16:58:20 -0700 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <49CC0B47.6000508@gmx.net> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> Message-ID: <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> On Thu, Mar 26, 2009 at 4:09 PM, Peter P GMX wrote: > Hello Michael, > > I tried this, but received the same behaviour. It does not ask for the > defined PIN. Just curious - where do you define the PIN for this conference? -MC From asannucci at gmail.com Thu Mar 26 19:20:51 2009 From: asannucci at gmail.com (Andrea) Date: Thu, 26 Mar 2009 21:20:51 -0500 Subject: [Freeswitch-users] Error Compiling iksemel... References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com><1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> Message-ID: <1B117EFC18604C6E82663FDAEB3342A8@quos> try to install gnutls-devel (name on centos) before to compile. From kawarod at laposte.net Fri Mar 27 00:40:11 2009 From: kawarod at laposte.net (rod) Date: Fri, 27 Mar 2009 11:40:11 +0400 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 Message-ID: <49CC82DB.1000900@laposte.net> Hi, I did some tests with FS to transcode SIP INFO to RFC2833 (and vice versa) and it's working fine when FS stays in the media path with default configuration. But my setup is the following: - Core network requires SIP INFO - Peerings require RFC2833 all would be fine with FS if my SIP Peers were not enforcing G729 (discarding G711) so that I have to use the directive in my dialplan cause FS can't deal with G729 except in pass-through. It's sad, but G729 is still a reality in Telco World. So do you think there could be a way to deal with DTMF even if not analyzing RTP for transcoding. My commercial SBC is doing this, but it sucks and that's the last step before final migration to FS. regards, rod From jason at jasonjgw.net Fri Mar 27 00:58:20 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 27 Mar 2009 18:58:20 +1100 Subject: [Freeswitch-users] Multiple calls with PortAudio In-Reply-To: <20090326043931.GA6652@jdc.jasonjgw.net> References: <20090326043931.GA6652@jdc.jasonjgw.net> Message-ID: <20090327075820.GA15756@jdc.jasonjgw.net> While I was trying to obtain more detailed logs of my portaudio problems, FreeSWITCH crashed, leaving a core file. The backtraces are here: http://pastebin.freeswitch.org/7998 As far as I can remember, at the time of the segfault, one channel was trying to connect and not succeeding; I had just issued a pa hangup command on it and then a pa call to try connecting again. Since my memory of exactly what was happening isn't as reliable as it should be, the value of the backtraces may be diminished. As to the portaudio problem, with rev. 12701 (Debian Sid, kernel 2.6.29, x86_64 architecture), the situation appears to be that the second and subsequent concurrent portaudio calls sometimes wait for a long time after issuing a log message such as the following: [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel sofia/internal/1000 at 192.168.0.2:5070 [d6e56642-1a9b-11de-b23e-c5a9450df57d] These calls do not always complete successfully, but I'm still trying to collect more precise details of when and why they fail. With apologies for being unable to use Jira, if anything valuable appears in the backtraces, you are welcome to let me know via the list or by e-mail. I have previously seen crashes while working with multiple portaudio calls, but I don't yet have a reliable means to reproduce them. If the backtraces are revealing, that's good, but if not, that's fine too and I'll collect better particulars next time. From Prometheus001 at gmx.net Fri Mar 27 01:27:42 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Mar 2009 09:27:42 +0100 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> Message-ID: <49CC8DFE.3050104@gmx.net> It's defined via XML-Curl, and manual dialling and transfering do trigger the same xml-curl request. This means that this conference number is not defined in the any xml conf file. If I transfer a call (without PIN) and then manually dial with another phone into this conf with PIN, both calls are in the same conference. I have SVN rev 12796. Best regards Peter Michael Collins schrieb: > On Thu, Mar 26, 2009 at 4:09 PM, Peter P GMX wrote: > >> Hello Michael, >> >> I tried this, but received the same behaviour. It does not ask for the >> defined PIN. >> > > Just curious - where do you define the PIN for this conference? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ttroy50 at gmail.com Fri Mar 27 02:55:04 2009 From: ttroy50 at gmail.com (Thomas Troy) Date: Fri, 27 Mar 2009 09:55:04 +0000 Subject: [Freeswitch-users] sipp emulating a registered end point In-Reply-To: References: Message-ID: When I need to do something like this, what I do is set sipp2 to have 2 scripts both using the same IP and port. One sends the register and deals with authentication \ OK. The other is then run after this to wait and receive the incoming call. So you would run them like sipp2_register - Performs registration and ends sipp2_receiveCall - Waits for incoming call, while listening on same IP \ Port as sipp2_register sipp1_makeCall - Makes outgoing call A sample register scenario is From: TestUser1 ;tag=[pid][call_number] Call-ID: [call_id] CSeq: 1 REGISTER Contact: Expires: 240 User-Agent: SIPp Content-Length: 0 ]]> From: TestUser1 ;tag=[pid][call_number] Call-ID: [call_id] CSeq: 2 REGISTER Contact: Expires: 240 User-Agent: SIPp [authentication username=user password=pass] Content-Length: 0 ]]> 2009/3/26 Jonas Gauffin > Hello > I want to achive this: Sipp1 -> FS -> Sipp2 > > Sipp1 emulates a inbound calls (easy to achive) > Sipp2 should emulate a registered user (i.e. register with FS and then just > wait for calls and hangup when sipp1 hangsup) > > How do I configure sipp as "Sipp2"? > > Thanks, > Jonas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/4a25a7c5/attachment-0002.html From mattdfong at gmail.com Fri Mar 27 04:48:45 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 27 Mar 2009 18:48:45 +0700 Subject: [Freeswitch-users] rubymod - ESL compile error Message-ID: <4256bf830903270448g5edc1a22sba2a91ea85fdbc1b@mail.gmail.com> I'm trying to get rubymod, working to experiment with it, but I'm getting the following error when I try to make on my Ubuntu system. root at ubuntu:/usr/src/freeswitch/libs/esl# make rubymod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C ruby make[1]: Entering directory `/usr/src/freeswitch/libs/esl/ruby' g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -lruby -o ESL.so -L. /usr/bin/ld: cannot find -lruby collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/ruby' make: *** [rubymod] Error 2 I'm currently using event sockets with a fully ruby implementation, but it's sort of slow at reading sockets. If I can get it working, it will be interesting seeing if I can improve performance. Does rubymod support events the same way the perlmod does? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/6fa833b2/attachment-0002.html From mattdfong at gmail.com Fri Mar 27 05:12:50 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 27 Mar 2009 19:12:50 +0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based Message-ID: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> I've been playing around with using freeswitch.EventConsumer in a lua process that starts-up when FS boots, and stays in the background. I've setup the example on the wiki, but the example uses session:execute("sleep",1000), and essentially loops every second until an event is fired. I'm wondering if there is a more event-driven way to accomplish this? I tried asking for help in #lua, but they said the project (FS) needed to implement event-driven programming for this to work. To me, it seems sort of silly to implement freeswitch.EventConsumer without a way for it to be executed event-wise Is using lua ESL the only option? There isn't any lua example scripts in libs/esl/lua to demonstrate how to handle events. if mod_lua can't handle events, can the mod_javascript utilize it? Thanks. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/024a368a/attachment-0002.html From andy at fabulous4.co.uk Fri Mar 27 05:34:46 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Fri, 27 Mar 2009 12:34:46 -0000 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: <2365564C-92A8-483E-9FCB-34EBE71EC256@avgs.ca> Message-ID: Thanks for your help folks, the ping parameter seems to have resolved the gateway connection issue but I now seem to be having a related issue with calls being cut off after a number of seconds. The freeswitch logs show a normal call clearing. I am indeed behind a NAT firewall which I'm assuming is the main issue. do you have any further tips to make this more stable and prevent the call cut off? Many thanks Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: 18 March 2009 14:46 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Losing Gateway registration if you are behind NAT it is possible that your router "forgot" the mapping betweeen FS and your provider, try adding to your gateway. Math On 18-Mar-09, at 10:07 AM, Brian West wrote: Upgrade to 1.03 or SVN Trunk /b On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: Hi, I've recently ugrade to version 1.02 of freeswitch and am having some problems with my gateway registrations. The gateway successfully registers with my voip provider when freeswitch first starts but if left running it seems to loose it's connection to my voip provider. I can get it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/b5877b66/attachment-0002.html From brian at freeswitch.org Fri Mar 27 06:50:30 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 08:50:30 -0500 Subject: [Freeswitch-users] Multiple calls with PortAudio In-Reply-To: <20090327075820.GA15756@jdc.jasonjgw.net> References: <20090326043931.GA6652@jdc.jasonjgw.net> <20090327075820.GA15756@jdc.jasonjgw.net> Message-ID: <9D237452-A946-492C-ABC2-FEDDF453E9AF@freeswitch.org> Please direct the report to http://jira.freeswitch.org /b On Mar 27, 2009, at 2:58 AM, Jason White wrote: > While I was trying to obtain more detailed logs of my portaudio > problems, > FreeSWITCH crashed, leaving a core file. > > The backtraces are here: > http://pastebin.freeswitch.org/7998 Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/1fd06fd5/attachment-0002.html From brian at freeswitch.org Fri Mar 27 06:49:09 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 08:49:09 -0500 Subject: [Freeswitch-users] rubymod - ESL compile error In-Reply-To: <4256bf830903270448g5edc1a22sba2a91ea85fdbc1b@mail.gmail.com> References: <4256bf830903270448g5edc1a22sba2a91ea85fdbc1b@mail.gmail.com> Message-ID: <763DC25A-FECF-4E40-9089-4C8CCC5B8943@freeswitch.org> http://jira.freeswitch.org/browse/ESL-7 I think this might apply to you. /b On Mar 27, 2009, at 6:48 AM, Matthew Fong wrote: > /usr/bin/ld: cannot find -lruby > collect2: ld returned 1 exit status Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/4ef56170/attachment-0002.html From jonas.gauffin at gmail.com Fri Mar 27 07:05:13 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Fri, 27 Mar 2009 15:05:13 +0100 Subject: [Freeswitch-users] sipp emulating a registered end point In-Reply-To: References: Message-ID: thanks! 2009/3/27 Thomas Troy > When I need to do something like this, what I do is set sipp2 to have 2 > scripts both using the same IP and port. > > One sends the register and deals with authentication \ OK. > The other is then run after this to wait and receive the incoming call. > > So you would run them like > > sipp2_register - Performs registration and ends > sipp2_receiveCall - Waits for incoming call, while listening > on same IP \ Port as sipp2_register > sipp1_makeCall - Makes outgoing call > > > A sample register scenario is > > > > > > REGISTER sip:[remote_ip]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] > [local_ip]:[local_port];branch=[branch]1;rport > Max-Forwards: 70 > To: TestUser1 > From: TestUser1 [remote_ip]:[remote_port]>;tag=[pid][call_number] > Call-ID: [call_id] > CSeq: 1 REGISTER > Contact: > Expires: 240 > User-Agent: SIPp > Content-Length: 0 > > ]]> > > > > > > > > REGISTER sip:[remote_ip]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] > [local_ip]:[local_port];branch=[branch]2;rport > Max-Forwards: 70 > To: TestUser1 > From: TestUser1 [remote_ip]:[remote_port]>;tag=[pid][call_number] > Call-ID: [call_id] > CSeq: 2 REGISTER > Contact: > Expires: 240 > User-Agent: SIPp > [authentication username=user password=pass] > Content-Length: 0 > > ]]> > > > > > > > > > > > > > > > 2009/3/26 Jonas Gauffin > >> Hello >> I want to achive this: Sipp1 -> FS -> Sipp2 >> >> Sipp1 emulates a inbound calls (easy to achive) >> Sipp2 should emulate a registered user (i.e. register with FS and then >> just wait for calls and hangup when sipp1 hangsup) >> >> How do I configure sipp as "Sipp2"? >> >> Thanks, >> Jonas >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/03dc6d51/attachment-0002.html From brian at freeswitch.org Fri Mar 27 07:18:27 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 09:18:27 -0500 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <49CC82DB.1000900@laposte.net> References: <49CC82DB.1000900@laposte.net> Message-ID: <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> On Mar 27, 2009, at 2:40 AM, rod wrote: > Hi, > > I did some tests with FS to transcode SIP INFO to RFC2833 (and vice > versa) and it's working fine when FS stays in the media path with > default configuration. > > But my setup is the following: > - Core network requires SIP INFO > - Peerings require RFC2833 > > all would be fine with FS if my SIP Peers were not enforcing G729 > (discarding G711) so that I have to use the directive application="set" data="proxy_media=true"/> in my dialplan cause FS > can't deal with G729 except in pass-through. Can't use proxy media in this case. (I highly recommend you not use Proxy Media mode) > > It's sad, but G729 is still a reality in Telco World. Coming soon! > > So do you think there could be a way to deal with DTMF even if not > analyzing RTP for transcoding. My commercial SBC is doing this, but it > sucks and that's the last step before final migration to FS. > > regards, > rod Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/09438dc8/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 27 07:18:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 09:18:10 -0500 Subject: [Freeswitch-users] Multiple calls with PortAudio In-Reply-To: <20090327075820.GA15756@jdc.jasonjgw.net> References: <20090326043931.GA6652@jdc.jasonjgw.net> <20090327075820.GA15756@jdc.jasonjgw.net> Message-ID: <191c3a030903270718k4d5dbdfvd19e57b5bad381cf@mail.gmail.com> Are you updating with "make current" each time? On Fri, Mar 27, 2009 at 2:58 AM, Jason White wrote: > While I was trying to obtain more detailed logs of my portaudio problems, > FreeSWITCH crashed, leaving a core file. > > The backtraces are here: > http://pastebin.freeswitch.org/7998 > > As far as I can remember, at the time of the segfault, one channel was > trying > to connect and not succeeding; I had just issued a pa hangup command on it > and > then a pa call to try connecting again. Since my memory of exactly what was > happening isn't as reliable as it should be, the value of the backtraces > may > be diminished. > > As to the portaudio problem, with rev. 12701 (Debian Sid, kernel 2.6.29, > x86_64 architecture), the situation appears to be that the second and > subsequent concurrent portaudio calls sometimes wait for a long time after > issuing a log message such as the following: > > [NOTICE] switch_channel.c:597 switch_channel_set_name() New Channel > sofia/internal/1000 at 192.168.0.2:5070[d6e56642-1a9b-11de-b23e-c5a9450df57d] > > These calls do not always complete successfully, but I'm still trying to > collect more precise details of when and why they fail. > > With apologies for being unable to use Jira, if anything valuable appears > in > the backtraces, you are welcome to let me know via the list or by e-mail. > > I have previously seen crashes while working with multiple portaudio calls, > but I don't yet have a reliable means to reproduce them. > > If the backtraces are revealing, that's good, but if not, that's fine too > and > I'll collect better particulars next time. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/52a6ca02/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 27 07:52:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 09:52:48 -0500 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> Message-ID: <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> Sort of silly?, I am not sure what you are talking about. I t's called *event*Consumer right? what do you mean by event based? There is no need to create a session? con = freeswitch.EventConsumer("all"); now you have a consumer obj every time you call con:pop() with no arg you will either get an event or nil when there are no events to consume. every time you call con:pop(1) the consumer object will block until there is an event. So you use the first way in conjunction with some other lock to do async or the 2nd way you do a dedicated blocking loop. I don't know what you said in #lua but, umm duhhhh I think we have an event driven programming under control..... We have a dedicated eventing engine in the core with scaling backend dispatcher threads that can handle hundereds of thousand of events at a time. There is also Event Socket (the word *event* again) that can connect to tcp and listen for *events* you can also write your code in C with the trivial module API that allows you to bind to an event internally and pretty much do whatever you want. 2009/3/27 Matthew Fong > I've been playing around with using freeswitch.EventConsumer in a lua > process that starts-up when FS boots, and stays in the background. I've > setup the example on the wiki, but the example uses > session:execute("sleep",1000), and essentially loops every second until an > event is fired. I'm wondering if there is a more event-driven way to > accomplish this? > I tried asking for help in #lua, but they said the project (FS) needed to > implement event-driven programming for this to work. To me, it seems sort of > silly to implement freeswitch.EventConsumer without a way for it to be > executed event-wise > > Is using lua ESL the only option? There isn't any lua example scripts in > libs/esl/lua to demonstrate how to handle events. > > if mod_lua can't handle events, can the mod_javascript utilize it? Thanks. > > --matt > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/451757a5/attachment-0002.html From kawarod at laposte.net Fri Mar 27 08:07:51 2009 From: kawarod at laposte.net (rod) Date: Fri, 27 Mar 2009 19:07:51 +0400 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> References: <49CC82DB.1000900@laposte.net> <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> Message-ID: <49CCEBC7.1070807@laposte.net> Hi Brian, don't understand very well your advice: --> Can't use proxy media in this case. (I highly recommend you not use Proxy Media mode) If i want to hide my topology network and deal with G729, I must use proxy media ? Why is Proxy media mode not recommended ?? regards. rod Brian West wrote: > > On Mar 27, 2009, at 2:40 AM, rod wrote: > >> Hi, >> >> I did some tests with FS to transcode SIP INFO to RFC2833 (and vice >> versa) and it's working fine when FS stays in the media path with >> default configuration. >> >> But my setup is the following: >> - Core network requires SIP INFO >> - Peerings require RFC2833 >> >> all would be fine with FS if my SIP Peers were not enforcing G729 >> (discarding G711) so that I have to use the directive > application="set" data="proxy_media=true"/> in my dialplan cause FS >> can't deal with G729 except in pass-through. > > Can't use proxy media in this case. (I highly recommend you not use > Proxy Media mode) > >> >> It's sad, but G729 is still a reality in Telco World. > > Coming soon! > >> >> So do you think there could be a way to deal with DTMF even if not >> analyzing RTP for transcoding. My commercial SBC is doing this, but it >> sucks and that's the last step before final migration to FS. >> >> regards, >> rod > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From helmut.kuper at ewetel.de Fri Mar 27 08:09:09 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 27 Mar 2009 16:09:09 +0100 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CB9E0C.4030300@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> <49CB9E0C.4030300@ewetel.de> Message-ID: <49CCEC15.8010500@ewetel.de> Hello, today I killed that special thread via "kill -9" a simple kill didn't helped. Unfortunately this led to a normal shutdown of FS although I killed not the parent process. :( After restart of FS the server has a normal load again. regards and a nice weekend Helmut From anthony.minessale at gmail.com Fri Mar 27 08:13:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 10:13:59 -0500 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <49CCEBC7.1070807@laposte.net> References: <49CC82DB.1000900@laposte.net> <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> <49CCEBC7.1070807@laposte.net> Message-ID: <191c3a030903270813x1319825bva59b09b789f79855@mail.gmail.com> if you enable mod_g729 you can use freeswitch normally with that g729 codec as long as no transcoding is enabled (same passthru concept as proxy_media_mode) On Fri, Mar 27, 2009 at 10:07 AM, rod wrote: > Hi Brian, > > don't understand very well your advice: > --> Can't use proxy media in this case. (I highly recommend you not use > Proxy Media mode) > > If i want to hide my topology network and deal with G729, I must use > proxy media ? > Why is Proxy media mode not recommended ?? > > regards. > rod > > > > Brian West wrote: > > > > On Mar 27, 2009, at 2:40 AM, rod wrote: > > > >> Hi, > >> > >> I did some tests with FS to transcode SIP INFO to RFC2833 (and vice > >> versa) and it's working fine when FS stays in the media path with > >> default configuration. > >> > >> But my setup is the following: > >> - Core network requires SIP INFO > >> - Peerings require RFC2833 > >> > >> all would be fine with FS if my SIP Peers were not enforcing G729 > >> (discarding G711) so that I have to use the directive >> application="set" data="proxy_media=true"/> in my dialplan cause FS > >> can't deal with G729 except in pass-through. > > > > Can't use proxy media in this case. (I highly recommend you not use > > Proxy Media mode) > > > >> > >> It's sad, but G729 is still a reality in Telco World. > > > > Coming soon! > > > >> > >> So do you think there could be a way to deal with DTMF even if not > >> analyzing RTP for transcoding. My commercial SBC is doing this, but it > >> sucks and that's the last step before final migration to FS. > >> > >> regards, > >> rod > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/8c4491f2/attachment-0002.html From jht at lj.net Fri Mar 27 03:11:45 2009 From: jht at lj.net (James H Thompson) Date: Fri, 27 Mar 2009 00:11:45 -1000 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls Message-ID: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> I need to generate calls with Invite URIs in this format: INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 SIP/2.0 Is there an easy way to do this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/42a5026e/attachment-0002.html From fdelawarde at wirelessmundi.com Fri Mar 27 05:05:04 2009 From: fdelawarde at wirelessmundi.com (Francois Delawarde) Date: Fri, 27 Mar 2009 13:05:04 +0100 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues Message-ID: <1238155504.4364.222.camel@localhost.localdomain> Hello and welcome me into FreeSWITCH's world! <= sorry that was rude I am (hoping to say "I was" soon) a heavy user of Asterisk's call queues for small call centers with sometimes empty queues and all agents idle for a few seconds. FreeSWITCH's mod_fifo algorithm is apparently quite different than Asterisk's app_queue. Instead of choosing an agent for a each call once it gets to the bottom of the queue given a specific strategy, FreeSWITCH does the inverse and finds a call once an agent is free given a strategy (the call that has waited longer from all the agent's queues, or the call in the queue that currently has more calls waiting). Am I right? If the above deduction is correct, while it seems a MUCH better choice for heavier call centers that always have calls in their queues ("in queue" calls are not delayed by the processing of the call at the end of the queue), I have a few doubts for what would happen in small call centers when those queues sometimes get empty and several agents "fight" for the incoming calls. My questions are following: - If for example 4 agents are "connected" (fifo out) to an empty queue, what happens when a call arrives? Do the 4 agents ring? If not, how do we know which agent get the call? - Is there an [easy] way (with some javascript or similar) to "emulate" Asterisk's distribution strategies to agents (by amount of time without calls, total number of answered calls, round robing, ...) in this cases? A couple of other newbie questions that has nothing to do with the above: - Is there a way to execute some PHP scripts for each call that would do the bridging or call applications (like Asterisk's AGI)? - What is the recommended language for features, speed, and programming ease (not a priority) for this kind of scripts (C? LUA?, Javascript?, ..)? Thanks in advance, Fran?ois. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/98fbb298/attachment-0002.html From mike at jerris.com Fri Mar 27 08:17:56 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Mar 2009 11:17:56 -0400 Subject: [Freeswitch-users] Losing Gateway registration In-Reply-To: References: Message-ID: <6E440475-070D-4C39-B5A3-AC91781B4555@jerris.com> Look closer at the logs or sip trace, this sounds like a failed session timer to me. Mike On Mar 27, 2009, at 8:34 AM, Andy Ayers wrote: > Thanks for your help folks, the ping parameter seems to have > resolved the gateway connection issue but I now seem to be having a > related issue with calls being cut off after a number of seconds. > The freeswitch logs show a normal call clearing. I am indeed behind > a NAT firewall which I'm assuming is the main issue. do you have any > further tips to make this more stable and prevent the call cut off? > > Many thanks > Andy > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Mathieu Rene > Sent: 18 March 2009 14:46 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Losing Gateway registration > > if you are behind NAT it is possible that your router "forgot" the > mapping betweeen FS and your provider, try adding name="ping" value="30" /> to your gateway. > > Math > > On 18-Mar-09, at 10:07 AM, Brian West wrote: > >> Upgrade to 1.03 or SVN Trunk >> >> /b >> >> On Mar 18, 2009, at 6:20 AM, Andy Ayers wrote: >> >>> Hi, >>> >>> I've recently ugrade to version 1.02 of freeswitch and am having >>> some problems with my gateway registrations. The gateway >>> successfully registers with my voip provider when freeswitch first >>> starts but if left running it seems to loose it's connection to my >>> voip provider. I can get it to reconnect with a sofia restart. I'm >>> using the same provider and user account as with the old version >>> of the software. Can you suggest any reaosn why this may be >>> happening and how I can prevent it? >>> >>> Many thanks >>> Andy >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/09b10649/attachment-0002.html From brian at freeswitch.org Fri Mar 27 08:23:17 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 10:23:17 -0500 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> Message-ID: <09BC8800-D604-40C3-B682-43DFF20DF3B0@freeswitch.org> Params are not added before the @ sign that I'm pretty sure of. Results in INVITE sip:blah at blah;this=kewl;that=iskewl SIP/2.0 You also have sip_invite_to_params, sip_invite_from_params, sip_invite_contact_params /b On Mar 27, 2009, at 5:11 AM, James H Thompson wrote: > I need to generate calls with Invite URIs in this format: > > INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 > SIP/2.0 > > Is there an easy way to do this? > > Thanks. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/442a3579/attachment-0002.html From mike at jerris.com Fri Mar 27 08:32:37 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Mar 2009 11:32:37 -0400 Subject: [Freeswitch-users] rubymod - ESL compile error In-Reply-To: <763DC25A-FECF-4E40-9089-4C8CCC5B8943@freeswitch.org> References: <4256bf830903270448g5edc1a22sba2a91ea85fdbc1b@mail.gmail.com> <763DC25A-FECF-4E40-9089-4C8CCC5B8943@freeswitch.org> Message-ID: <28E9E7EB-7F31-4CEE-89EC-5348CC055E1D@jerris.com> fixed revision 12805. Mike On Mar 27, 2009, at 9:49 AM, Brian West wrote: > http://jira.freeswitch.org/browse/ESL-7 > > I think this might apply to you. > > /b > > On Mar 27, 2009, at 6:48 AM, Matthew Fong wrote: > >> /usr/bin/ld: cannot find -lruby >> collect2: ld returned 1 exit status > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/19328339/attachment-0002.html From mike at jerris.com Fri Mar 27 08:41:46 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Mar 2009 11:41:46 -0400 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> Message-ID: You seem to be confusing your standards, those 2 specs are about tel: uri's not sip: uris. Sending a tel uri I am not sure we can do, where would we send it to? Mike On Mar 27, 2009, at 6:11 AM, James H Thompson wrote: > I need to generate calls with Invite URIs in this format: > > INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 > SIP/2.0 > > Is there an easy way to do this? > > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/d8698a1b/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 27 08:55:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 10:55:46 -0500 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues In-Reply-To: <1238155504.4364.222.camel@localhost.localdomain> References: <1238155504.4364.222.camel@localhost.localdomain> Message-ID: <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> 2009/3/27 Francois Delawarde > Hello and welcome me into FreeSWITCH's world! <= sorry that was rude > > I am (hoping to say "I was" soon) a heavy user of Asterisk's call queues > for small call centers with sometimes empty queues and all agents idle for a > few seconds. > > FreeSWITCH's mod_fifo algorithm is apparently quite different than > Asterisk's app_queue. Instead of choosing an agent for a each call once it > gets to the bottom of the queue given a specific strategy, FreeSWITCH does > the inverse and finds a call once an agent is free given a strategy (the > call that has waited longer from all the agent's queues, or the call in the > queue that currently has more calls waiting). Am I right? > > If the above deduction is correct, while it seems a MUCH better choice for > heavier call centers that always have calls in their queues ("in queue" > calls are not delayed by the processing of the call at the end of the > queue), I have a few doubts for what would happen in small call centers when > those queues sometimes get empty and several agents "fight" for the incoming > calls. My questions are following: > > - If for example 4 agents are "connected" (fifo out) to an empty queue, > what happens when a call arrives? Do the 4 agents ring? If not, how do we > know which agent get the call? > If you are using on-hook agents, it will place as many outbound calls as there are people waiting. If you are using off-hook agents it will just connect the first free agent. > - Is there an [easy] way (with some javascript or similar) to "emulate" > Asterisk's distribution strategies to agents (by amount of time without > calls, total number of answered calls, round robing, ...) in this cases? > Easiest way would be to write a patch in C to mod_fifo it'self or propose a bounty for features and see if you can get the change approved by the developers. > > A couple of other newbie questions that has nothing to do with the above: > - Is there a way to execute some PHP scripts for each call that would do > the bridging or call applications (like Asterisk's AGI)? > Your best bet would be to not try to do anything "like asterisk" FreeSWITCH is a paradigm shift from asterisk and you may defeat yourself trying to do anything the same way. That said, yes, look at Event Socket and ESL, (using asterisk terminology, it's a combination of AGI and manager). > - What is the recommended language for features, speed, and programming > ease (not a priority) for this kind of scripts (C? LUA?, Javascript?, ..)? > C > > Thanks in advance, > Fran?ois. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/09af9a02/attachment-0002.html From dujinfang at gmail.com Fri Mar 27 09:12:17 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 28 Mar 2009 00:12:17 +0800 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> Message-ID: <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: > We do not support ubuntu interpid, it has at least 3 known fatal > issues not experienced by all but nonetheless enough to make us > unwilling to support it. I use Ubuntu gutsy in production and interipid in test env. It works well. Can you briefly explain the 3 fatal issues Anthony? It will help me know potential risks. > > > It's "use at your own risk" or use the stable branch "hardy" for any > support. > > > > On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds > wrote: > Has there been any progress getting FreeSWITCH to build on Ubuntu > Intrepid without downgrading libtool? > I successfully built FS on intrepid. I simply done this by changing the apt-source to Hardy and installed libtool. Obviously I changed the apt-source back to intrepid after I installed libtool. And, another approach. Install libtool from source should be as easy as configure && make && make install. I done this on a new CentOS4 because the default yum install of libtool on CentOS4 is old than FS required. > Thanks! > > Sincerely, > Trevor Hammonds > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/a68ea737/attachment-0002.html From kawarod at laposte.net Fri Mar 27 09:23:59 2009 From: kawarod at laposte.net (rod) Date: Fri, 27 Mar 2009 20:23:59 +0400 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <191c3a030903270813x1319825bva59b09b789f79855@mail.gmail.com> References: <49CC82DB.1000900@laposte.net> <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> <49CCEBC7.1070807@laposte.net> <191c3a030903270813x1319825bva59b09b789f79855@mail.gmail.com> Message-ID: <49CCFD9F.70600@laposte.net> Hello, I have this error when not enablig proxy_media: 2009-03-27 19:54:44 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-27 19:54:44 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! Sure there is an option to check. Any pointers. regards. Anthony Minessale wrote: > if you enable mod_g729 you can use freeswitch normally with that g729 > codec as long > as no transcoding is enabled (same passthru concept as proxy_media_mode) > > > On Fri, Mar 27, 2009 at 10:07 AM, rod > wrote: > > Hi Brian, > > don't understand very well your advice: > --> Can't use proxy media in this case. (I highly recommend you > not use > Proxy Media mode) > > If i want to hide my topology network and deal with G729, I must use > proxy media ? > Why is Proxy media mode not recommended ?? > > regards. > rod > > > > Brian West wrote: > > > > On Mar 27, 2009, at 2:40 AM, rod wrote: > > > >> Hi, > >> > >> I did some tests with FS to transcode SIP INFO to RFC2833 (and vice > >> versa) and it's working fine when FS stays in the media path with > >> default configuration. > >> > >> But my setup is the following: > >> - Core network requires SIP INFO > >> - Peerings require RFC2833 > >> > >> all would be fine with FS if my SIP Peers were not enforcing G729 > >> (discarding G711) so that I have to use the directive >> application="set" data="proxy_media=true"/> in my dialplan cause FS > >> can't deal with G729 except in pass-through. > > > > Can't use proxy media in this case. (I highly recommend you not use > > Proxy Media mode) > > > >> > >> It's sad, but G729 is still a reality in Telco World. > > > > Coming soon! > > > >> > >> So do you think there could be a way to deal with DTMF even if not > >> analyzing RTP for transcoding. My commercial SBC is doing this, > but it > >> sucks and that's the last step before final migration to FS. > >> > >> regards, > >> rod > > > > Brian West > > brian at freeswitch.org > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Fri Mar 27 09:25:06 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Mar 2009 17:25:06 +0100 Subject: [Freeswitch-users] condition matching on variables which have been set in the dialplan Message-ID: <49CCFDE2.4050704@gmx.net> I try to use speed dialling and masked numbers in a dialplan through xml-curl. For the XML I use templates which I fill with variables. The numbering plan is set up in a way that any number can be a speed dialling or masked number, so I cannot parse them via Regex in the XML part of the dialplan. E.g. * 12345 is a normal phone * 12346 is a speed dialling number => 0049xxxxxxxxxx * 12347 is a normal phone * 4 is a speed dialling number => 0049xxxxxxxxxx So I need to substitute a variable with the final number to be dialled. This final number then needs to be parsed in the dialplan to indentify how to handle it (bridge, conference, voicemail etc.) I have special reasons to do that, so please do not ask me why. So the dialplan is as following . . . . . . . . In the first condition I set the substituted final destination number. This is dynamically substituted in the template in my application via xml-curl dependend on which kind of number is dialled. In this case a German number is substituted. In the following conditions I would like to set the gateways. What is happening in the logs? * I dial e.g. "12346" for a speed dialling number * The first condition is parsed correctly, and the variables are set (Action set(destination_number=0049xxxxxxxxxxxx) * in the second condition "${variable_destination_number} is not set to the new value. It's still "12346".(I also tried conditions based on "${destination_number}" and "destination_number"). * In the logs the execution of "set" and "export" in fact is shown the whole conditions are parsed. Also the info application is outputted after all conditions are parsed. E.g. EXECUTE sofia/internal/10000 at sip.domain.de set(destination_number=0049xxxxxxxxxxxx) * the "info" app shows me that "variable_destination_number" is set to the right number, but it seems to be too late? Question: Are these lines not handled sequentially (I am using a quad core machine)? Any other idea how to solve this? Best regards Peter From brian at freeswitch.org Fri Mar 27 09:55:08 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 11:55:08 -0500 Subject: [Freeswitch-users] condition matching on variables which have been set in the dialplan In-Reply-To: <49CCFDE2.4050704@gmx.net> References: <49CCFDE2.4050704@gmx.net> Message-ID: <7C44CF00-5B22-4E19-A0A8-74C820E02A46@freeswitch.org> Remember the dialplan is NOT executed when its parsed so you can't set a var then condition on that exact var on the next line.. that var doesn't exist. /b On Mar 27, 2009, at 11:25 AM, Peter P GMX wrote: > > expression="^[0-9]\d[0,16}$" > continue="true">. > data="destination_number=0049xxxxxxxxx"/>. > data="destination_number=0049xxxxxxxxx"/>. > > > expression="^(00[1-9]\d{4,13})$"> > data="effective_caller_id_number=unknown"/>. > data="effective_caller_id_name=unknown"/>. > data="sofia/gateway/QSC_DE/$1 at sip.qsc.de"/>. > > > . > . > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/6ca27cc8/attachment-0002.html From mike at jerris.com Fri Mar 27 10:31:09 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 27 Mar 2009 13:31:09 -0400 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> Message-ID: <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> Another example of a fatal issue was the optimizer in gcc was breaking openzap code even with -O2. Mike On Mar 27, 2009, at 12:12 PM, dujinfang wrote: > > On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: >> We do not support ubuntu interpid, it has at least 3 known fatal >> issues not experienced by all but nonetheless enough to make us >> unwilling to support it. > > I use Ubuntu gutsy in production and interipid in test env. It > works well. Can you briefly explain the 3 fatal issues Anthony? It > will help me know potential risks. >> >> It's "use at your own risk" or use the stable branch "hardy" for >> any support. >> >> On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds > > wrote: >> Has there been any progress getting FreeSWITCH to build on Ubuntu >> Intrepid without downgrading libtool? >> > > I successfully built FS on intrepid. I simply done this by changing > the apt-source to Hardy and installed libtool. Obviously I changed > the apt-source back to intrepid after I installed libtool. > > And, another approach. Install libtool from source should be as easy > as configure && make && make install. I done this on a new CentOS4 > because the default yum install of libtool on CentOS4 is old than FS > required. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/a332d915/attachment-0002.html From msc at freeswitch.org Fri Mar 27 10:43:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 27 Mar 2009 10:43:13 -0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> Message-ID: <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> > con = freeswitch.EventConsumer("all"); > > now you have a consumer obj > > every time you call con:pop() with no arg you will either get an event or > nil when there are no events to consume. > every time you call con:pop(1) the consumer object will block until there is > an event. > > So you use the first way in conjunction with some other lock to do async or > the 2nd way you do a dedicated blocking loop. FYI, I added this information to the wiki page for freeswitch.EventConsumer. -MC From gkuri at ieee.org Fri Mar 27 11:19:57 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 27 Mar 2009 11:19:57 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> Message-ID: <49CD18CD.5000704@ieee.org> regarding gcc compiler optimizations, are they generally compatible with FS or should they be removed or does the configure strip them out? just curious, as I run Gentoo and use such optimizations as "-march=nocona -O2 -pipe -fomit-frame-pointer" not sure if they break things or I should be removing them before compiling FS? Gabe Michael Jerris wrote: > Another example of a fatal issue was the optimizer in gcc was breaking > openzap code even with -O2. > > Mike > > On Mar 27, 2009, at 12:12 PM, dujinfang wrote: > >> >> On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: >>> We do not support ubuntu interpid, it has at least 3 known fatal >>> issues not experienced by all but nonetheless enough to make us >>> unwilling to support it. >> >> I use Ubuntu gutsy in production and interipid in test env. It works >> well. Can you briefly explain the 3 fatal issues Anthony? It will help >> me know potential risks. >>> >>> It's "use at your own risk" or use the stable branch "hardy" for any >>> support. >>> >>> On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds >>> > wrote: >>> >>> Has there been any progress getting FreeSWITCH to build on Ubuntu >>> Intrepid without downgrading libtool? >>> >> >> I successfully built FS on intrepid. I simply done this by changing >> the apt-source to Hardy and installed libtool. Obviously I changed the >> apt-source back to intrepid after I installed libtool. >> >> And, another approach. Install libtool from source should be as easy >> as configure && make && make install. I done this on a new CentOS4 >> because the default yum install of libtool on CentOS4 is old than FS >> required. > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri Mar 27 11:30:01 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 13:30:01 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <49CD18CD.5000704@ieee.org> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> Message-ID: Usually if you don't know what they do... then you shouldn't use them! ;) /b On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote: > regarding gcc compiler optimizations, are they generally compatible > with > FS or should they be removed or does the configure strip them out? > just > curious, as I run Gentoo and use such optimizations as "-march=nocona > -O2 -pipe -fomit-frame-pointer" > > not sure if they break things or I should be removing them before > compiling FS? > > Gabe Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/11f858c1/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 27 11:55:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 13:55:27 -0500 Subject: [Freeswitch-users] SIP INFO <-> RFC2833 In-Reply-To: <49CCFD9F.70600@laposte.net> References: <49CC82DB.1000900@laposte.net> <0B9FFE5F-C579-4EC1-ABD7-E32B47681BEE@freeswitch.org> <49CCEBC7.1070807@laposte.net> <191c3a030903270813x1319825bva59b09b789f79855@mail.gmail.com> <49CCFD9F.70600@laposte.net> Message-ID: <191c3a030903271155g751cd9cfk4aab42f60291095c@mail.gmail.com> you have to set disable-transcoding as well to avoid any transcoding situations On Fri, Mar 27, 2009 at 11:23 AM, rod wrote: > Hello, > > I have this error when not enablig proxy_media: > 2009-03-27 19:54:44 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-03-27 19:54:44 [ERR] switch_core_io.c:723 > switch_core_session_write_frame() Codec G.729 decoder error! > > Sure there is an option to check. Any pointers. > > regards. > > > > > Anthony Minessale wrote: > > if you enable mod_g729 you can use freeswitch normally with that g729 > > codec as long > > as no transcoding is enabled (same passthru concept as proxy_media_mode) > > > > > > On Fri, Mar 27, 2009 at 10:07 AM, rod > > wrote: > > > > Hi Brian, > > > > don't understand very well your advice: > > --> Can't use proxy media in this case. (I highly recommend you > > not use > > Proxy Media mode) > > > > If i want to hide my topology network and deal with G729, I must use > > proxy media ? > > Why is Proxy media mode not recommended ?? > > > > regards. > > rod > > > > > > > > Brian West wrote: > > > > > > On Mar 27, 2009, at 2:40 AM, rod wrote: > > > > > >> Hi, > > >> > > >> I did some tests with FS to transcode SIP INFO to RFC2833 (and > vice > > >> versa) and it's working fine when FS stays in the media path with > > >> default configuration. > > >> > > >> But my setup is the following: > > >> - Core network requires SIP INFO > > >> - Peerings require RFC2833 > > >> > > >> all would be fine with FS if my SIP Peers were not enforcing G729 > > >> (discarding G711) so that I have to use the directive > >> application="set" data="proxy_media=true"/> in my dialplan cause > FS > > >> can't deal with G729 except in pass-through. > > > > > > Can't use proxy media in this case. (I highly recommend you not > use > > > Proxy Media mode) > > > > > >> > > >> It's sad, but G729 is still a reality in Telco World. > > > > > > Coming soon! > > > > > >> > > >> So do you think there could be a way to deal with DTMF even if not > > >> analyzing RTP for transcoding. My commercial SBC is doing this, > > but it > > >> sucks and that's the last step before final migration to FS. > > >> > > >> regards, > > >> rod > > > > > > Brian West > > > brian at freeswitch.org > > > > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at 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 at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > 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:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/5d684059/attachment-0002.html From gkuri at ieee.org Fri Mar 27 11:56:07 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 27 Mar 2009 11:56:07 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> Message-ID: <49CD2147.2070305@ieee.org> I'm not asking what they do, I'm asking those more familiar with FS whether the optimization flags are too aggressive for FS. What do you guys (developers) normalize use, just your basic -march=i686 -pipe ? Gabe Brian West wrote: > Usually if you don't know what they do... then you shouldn't use them! ;) > > > /b > > On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote: > >> regarding gcc compiler optimizations, are they generally compatible with >> FS or should they be removed or does the configure strip them out? just >> curious, as I run Gentoo and use such optimizations as "-march=nocona >> -O2 -pipe -fomit-frame-pointer" >> >> not sure if they break things or I should be removing them before >> compiling FS? >> >> Gabe > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris at fowler.cc Fri Mar 27 12:01:34 2009 From: chris at fowler.cc (Chris Fowler) Date: Fri, 27 Mar 2009 12:01:34 -0700 Subject: [Freeswitch-users] DTMF Missing Digits Message-ID: <1238180494.11750.1307673017@webmail.messagingengine.com> >> Sent: Wednesday, March 25, 2009 12:43 btw you'll have to reinstall your phrase macros .... make vm-sync I think should do it if it doesn't let me know... we removed the 250ms sleeps and that was the problem which we fixed. << I re-did the macros; the only change I could detect was the elimination of the 250ms sleeps; and the change to: I'm running build 12782; should this have fixed it? If so, I will follow the bug reporting instructions you sent earlier. Thanks, Chris. Here's the errors caught today on my production system. 2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '1101' 2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '55' 2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '015' 2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '000' 2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '0000' 2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' 2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input '006' From brian at freeswitch.org Fri Mar 27 12:03:29 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 14:03:29 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <49CD2147.2070305@ieee.org> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> <49CD2147.2070305@ieee.org> Message-ID: <0F0D9025-63A1-499C-A9F7-E215BCA20CD8@freeswitch.org> We usually don't specify anything extra! /b On Mar 27, 2009, at 1:56 PM, Gabriel Kuri wrote: > I'm not asking what they do, I'm asking those more familiar with FS > whether the optimization flags are too aggressive for FS. What do you > guys (developers) normalize use, just your basic -march=i686 -pipe ? > > Gabe Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/374cb9c9/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 27 12:04:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 14:04:17 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> Message-ID: <191c3a030903271204s73b96a92k77780034bc7b42be@mail.gmail.com> 1) There is an incompatibility on the fake ncurses wrapper that causes an instant seg fault unless you install the real ncurses. 2) The bleeding edge GCC builds an openzap binary that crashes instantly with no explanation in the core file from a minimal -O2 (that's just the one copmiler bug that we know about for sure, like cock roaches, see one, there are probably 1000) 3) They upgraded to libtool 2.0 which builds binaries that will not start. (easier said than done to upgrade ours too as we have to make sure we work on *every* plarform and the upgrade to make it work would break other operating systems we support) Bottom line, it's not their fault or anything but the choice to use all brand new versions of everything under the sun is not a good idea for your server, it's great that we have bleeding edge stuff or we would not have anyone to test stuff, we have a similar group of people always running SVN trunk of the day. But it's hard to stabalize code when both your code and the OS may be unstable at the same time. There is a reason they call it bleeding vs stable, which one would you rather be if you were in the hospital. =D 2009/3/27 dujinfang > > On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: > > We do not support ubuntu interpid, it has at least 3 known fatal issues not > experienced by all but nonetheless enough to make us unwilling to support > it. > > > I use Ubuntu gutsy in production and interipid in test env. It works well. > Can you briefly explain the 3 fatal issues Anthony? It will help me know > potential risks. > > > > It's "use at your own risk" or use the stable branch "hardy" for any > support. > > > > On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds wrote: > >> Has there been any progress getting FreeSWITCH to build on Ubuntu >> Intrepid without downgrading libtool? >> >> > I successfully built FS on intrepid. I simply done this by changing the > apt-source to Hardy and installed libtool. Obviously I changed the > apt-source back to intrepid after I installed libtool. > > And, another approach. Install libtool from source should be as easy as > configure && make && make install. I done this on a new CentOS4 because the > default yum install of libtool on CentOS4 is old than FS required. > > > Thanks! >> >> Sincerely, >> Trevor Hammonds >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/e22607b1/attachment-0002.html From brian at freeswitch.org Fri Mar 27 12:04:26 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 14:04:26 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238180494.11750.1307673017@webmail.messagingengine.com> References: <1238180494.11750.1307673017@webmail.messagingengine.com> Message-ID: Right and that is the fix for this. If you have the sleep's in your phrase macro's remove them and use the pause= param... you shouldn't have any problems. /b On Mar 27, 2009, at 2:01 PM, Chris Fowler wrote: > > I re-did the macros; the only change I could detect was the > elimination > of the 250ms sleeps; and the change to: > > > I'm running build 12782; should this have fixed it? If so, I will > follow the bug reporting instructions you sent earlier. > > Thanks, Chris. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/f2a095a4/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 27 12:08:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 14:08:23 -0500 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <49CD2147.2070305@ieee.org> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> <49CD2147.2070305@ieee.org> Message-ID: <191c3a030903271208v36ffbc95md5311af35dce6e62@mail.gmail.com> We've made no attempts to add any optimization flags on unix to date. We use the defaults and always build debug binaries. When we get some spare time we might go back and turn them on but so far we don't have much of a need to. On Fri, Mar 27, 2009 at 1:56 PM, Gabriel Kuri wrote: > I'm not asking what they do, I'm asking those more familiar with FS > whether the optimization flags are too aggressive for FS. What do you > guys (developers) normalize use, just your basic -march=i686 -pipe ? > > Gabe > > > Brian West wrote: > > Usually if you don't know what they do... then you shouldn't use them! > ;) > > > > > > /b > > > > On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote: > > > >> regarding gcc compiler optimizations, are they generally compatible with > >> FS or should they be removed or does the configure strip them out? just > >> curious, as I run Gentoo and use such optimizations as "-march=nocona > >> -O2 -pipe -fomit-frame-pointer" > >> > >> not sure if they break things or I should be removing them before > >> compiling FS? > >> > >> Gabe > > > > Brian West > > brian at freeswitch.org > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/f2ebf3bd/attachment-0002.html From gkuri at ieee.org Fri Mar 27 12:11:44 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 27 Mar 2009 12:11:44 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <0F0D9025-63A1-499C-A9F7-E215BCA20CD8@freeswitch.org> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> <49CD2147.2070305@ieee.org> <0F0D9025-63A1-499C-A9F7-E215BCA20CD8@freeswitch.org> Message-ID: <49CD24F0.5090702@ieee.org> ok, thanks, that pretty much amounts to the gcc defaults. perhaps I should recompile FS with those defaults and see if the current jira I have open goes away ;) ... Gabe Brian West wrote: > We usually don't specify anything extra! > > /b > > On Mar 27, 2009, at 1:56 PM, Gabriel Kuri wrote: > >> I'm not asking what they do, I'm asking those more familiar with FS >> whether the optimization flags are too aggressive for FS. What do you >> guys (developers) normalize use, just your basic -march=i686 -pipe ? >> >> Gabe > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Fri Mar 27 12:12:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 14:12:06 -0500 Subject: [Freeswitch-users] DTMF Missing Digits In-Reply-To: <1238180494.11750.1307673017@webmail.messagingengine.com> References: <1238180494.11750.1307673017@webmail.messagingengine.com> Message-ID: <191c3a030903271212q270ee6cbv353781981cefb0f2@mail.gmail.com> You should file the bug with the guy who dreamed up RFC2833 ;) Did you provide the menu you are using and what you expect to happen? There are cases where the way you set it up could cause your problems. You also have to realize that dtmf over sip is one of the top 10 gripes ppl have with the protocol and you may have to carefully analyze your traffic since we have zero other complaints open about your problem other than the one from several months ago that brian told you about. 2009/3/27 Chris Fowler > >> > Sent: Wednesday, March 25, 2009 12:43 > > btw you'll have to reinstall your phrase macros .... make vm-sync I > think should do it if it doesn't let me know... we removed the 250ms > sleeps and that was the problem which we fixed. > << > > I re-did the macros; the only change I could detect was the elimination > of the 250ms sleeps; and the change to: > > > I'm running build 12782; should this have fixed it? If so, I will > follow the bug reporting instructions you sent earlier. > > Thanks, Chris. > > > > Here's the errors caught today on my production system. > > 2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '1101' > 2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '55' > 2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '015' > 2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '000' > 2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '0000' > 2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > 2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > 2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > 2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > 2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548 > switch_ivr_menu_execute() IVR menu 'rightscale_ivr' caught invalid input > '006' > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/612f907e/attachment-0002.html From gkuri at ieee.org Fri Mar 27 12:16:35 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 27 Mar 2009 12:16:35 -0700 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <191c3a030903271208v36ffbc95md5311af35dce6e62@mail.gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <9825A881-66EC-4940-AB3B-EE38D2E2F2FB@jerris.com> <49CD18CD.5000704@ieee.org> <49CD2147.2070305@ieee.org> <191c3a030903271208v36ffbc95md5311af35dce6e62@mail.gmail.com> Message-ID: <49CD2613.4000602@ieee.org> Thanks, Anthony. I guess that answers my question, the build process doesn't use whatever is configured locally on the system. Gabe Anthony Minessale wrote: > We've made no attempts to add any optimization flags on unix to date. > We use the defaults and always build debug binaries. > > When we get some spare time we might go back and turn them on but so far > we don't have much of a need to. > > > On Fri, Mar 27, 2009 at 1:56 PM, Gabriel Kuri > wrote: > > I'm not asking what they do, I'm asking those more familiar with FS > whether the optimization flags are too aggressive for FS. What do you > guys (developers) normalize use, just your basic -march=i686 -pipe ? > > Gabe > > > Brian West wrote: > > Usually if you don't know what they do... then you shouldn't use > them! ;) > > > > > > /b > > > > On Mar 27, 2009, at 1:19 PM, Gabriel Kuri wrote: > > > >> regarding gcc compiler optimizations, are they generally > compatible with > >> FS or should they be removed or does the configure strip them > out? just > >> curious, as I run Gentoo and use such optimizations as "-march=nocona > >> -O2 -pipe -fomit-frame-pointer" > >> > >> not sure if they break things or I should be removing them before > >> compiling FS? > >> > >> Gabe > > > > Brian West > > brian at freeswitch.org > > > > > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Prometheus001 at gmx.net Fri Mar 27 12:23:22 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 27 Mar 2009 20:23:22 +0100 Subject: [Freeswitch-users] condition matching on variables which have been set in the dialplan In-Reply-To: <7C44CF00-5B22-4E19-A0A8-74C820E02A46@freeswitch.org> References: <49CCFDE2.4050704@gmx.net> <7C44CF00-5B22-4E19-A0A8-74C820E02A46@freeswitch.org> Message-ID: <49CD27AA.4000503@gmx.net> OK, understood. I will do it in a different way then. Brian West schrieb: > Remember the dialplan is NOT executed when its parsed so you can't set > a var then condition on that exact var on the next line.. that var > doesn't exist. > > /b > > > On Mar 27, 2009, at 11:25 AM, Peter P GMX wrote: > >> >> > continue="true">. >> > data="destination_number=0049xxxxxxxxx"/>. >> > data="destination_number=0049xxxxxxxxx"/>. >> >> >> > Its ${destination_number} > >> expression="^(00[1-9]\d{4,13})$"> >> > data="effective_caller_id_number=unknown"/>. >> > data="effective_caller_id_name=unknown"/>. >> > data="sofia/gateway/QSC_DE/$1 at sip.qsc.de >> "/>. >> >> >> . >> . >> > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Mar 27 12:27:59 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 14:27:59 -0500 Subject: [Freeswitch-users] condition matching on variables which have been set in the dialplan In-Reply-To: <49CD27AA.4000503@gmx.net> References: <49CCFDE2.4050704@gmx.net> <7C44CF00-5B22-4E19-A0A8-74C820E02A46@freeswitch.org> <49CD27AA.4000503@gmx.net> Message-ID: <9019BDB0-C72F-465E-A616-B0B6CDAE9EA9@freeswitch.org> You can execute_extension to revisit the dialplan at a later time kinda like a macro. /b On Mar 27, 2009, at 2:23 PM, Peter P GMX wrote: > OK, understood. I will do it in a different way then. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/5acab493/attachment-0002.html From chris at fowler.cc Fri Mar 27 12:40:08 2009 From: chris at fowler.cc (Chris Fowler) Date: Fri, 27 Mar 2009 12:40:08 -0700 Subject: [Freeswitch-users] DTMF Missing Digits Message-ID: <1238182808.17773.1307679957@webmail.messagingengine.com> >> Did you provide the menu you are using and what you expect to happen? Here's the setup; Caller -> FlowRoute - > FreeSwitch >> B: Right and that is the fix for this. If you have the sleep's in your phrase macro's remove them and use the pause= param... you shouldn't have any problems. Still seeing multiple issues logged during ivr process for mis-interpreted DTMF. Here's today's list from our production server. 2009-03-27 06:38:59 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '1100' 2009-03-27 07:20:33 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 07:20:41 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 08:33:25 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 09:41:14 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '1101' 2009-03-27 09:41:19 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '55' 2009-03-27 09:41:33 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '015' 2009-03-27 10:13:15 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 10:13:22 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 10:13:50 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 10:13:59 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 10:14:11 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '0000' 2009-03-27 10:56:00 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 10:57:44 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 10:57:57 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 10:58:09 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 10:59:06 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 11:58:35 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '028' 2009-03-27 11:59:27 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '050' 2009-03-27 12:01:52 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 12:02:01 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '000' 2009-03-27 12:02:41 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' 2009-03-27 12:02:53 [DEBUG] switch_ivr_menu.c:548 switch_ivr_menu_execute() IVR menu 'main_ivr' caught invalid input '006' Any other debug I can capture to assist? Thanks, Chris. From kristian.kielhofner at gmail.com Fri Mar 27 12:47:02 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 27 Mar 2009 15:47:02 -0400 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <09BC8800-D604-40C3-B682-43DFF20DF3B0@freeswitch.org> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <09BC8800-D604-40C3-B682-43DFF20DF3B0@freeswitch.org> Message-ID: <2d9149cd0903271247p37ebf34dib906126b6a9d8d5c@mail.gmail.com> Some platforms add/use URI params in the user portion of the URI. I was just reminded of this this week with Global Crossing: sip:+18005551212;npdi=yes at 0.0.0.0 That is (of course) the number portability display indicator... ;) Anyways, I had to rewrite the URI using OpenSIPS to strip the ";npdi=yes". I imagine you could do the same but I'm not sure how in FreeSWITCH... 2009/3/27 Brian West : > Params are not added before the @ sign that I'm pretty sure of. > data="sip_invite_params=this=kewl;that=iskewl"/> > > > > Results in?INVITE sip:blah at blah;this=kewl;that=iskewl SIP/2.0 > > You also have sip_invite_to_params, sip_invite_from_params, > sip_invite_contact_params > /b > > > On Mar 27, 2009, at 5:11 AM, James H Thompson wrote: > > I need to generate calls with Invite URIs in this format: > > INVITE?sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060?SIP/2.0 > > Is there an easy way to do this? > > Thanks. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Fri Mar 27 12:48:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 27 Mar 2009 15:48:50 -0400 Subject: [Freeswitch-users] Contacting Callie Message-ID: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> What is the best way (if any) to contact "Callie" for custom prompt work? I can't seem to find much about her. Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Fri Mar 27 12:49:16 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 14:49:16 -0500 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <2d9149cd0903271247p37ebf34dib906126b6a9d8d5c@mail.gmail.com> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <09BC8800-D604-40C3-B682-43DFF20DF3B0@freeswitch.org> <2d9149cd0903271247p37ebf34dib906126b6a9d8d5c@mail.gmail.com> Message-ID: Thats about dumb... um I don't think we let you put things in there.. but it would only be a few lines to do it...blah rfc's suck! /b On Mar 27, 2009, at 2:47 PM, Kristian Kielhofner wrote: > Some platforms add/use URI params in the user portion of the URI. > > I was just reminded of this this week with Global Crossing: > > sip:+18005551212;npdi=yes at 0.0.0.0 > > That is (of course) the number portability display indicator... ;) > > Anyways, I had to rewrite the URI using OpenSIPS to strip the > ";npdi=yes". I imagine you could do the same but I'm not sure how in > FreeSWITCH. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/3829c479/attachment-0002.html From jht at lj.net Fri Mar 27 12:52:30 2009 From: jht at lj.net (James H Thompson) Date: Fri, 27 Mar 2009 09:52:30 -1000 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> Message-ID: <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> Calls would be sent to the IP address after the '@' in the URI. Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as the user part of a SIP URI. My example Invite URI is the way we are receiving traffic from some of the major telecom carriers. We would like be able to generate calls using the same formats. ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Friday, March 27, 2009 5:41 AM Subject: Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls You seem to be confusing your standards, those 2 specs are about tel: uri's not sip: uris. Sending a tel uri I am not sure we can do, where would we send it to? Mike On Mar 27, 2009, at 6:11 AM, James H Thompson wrote: I need to generate calls with Invite URIs in this format: INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 SIP/2.0 Is there an easy way to do this? Thanks. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/6b5775de/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 27 13:05:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 15:05:07 -0500 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> Message-ID: <191c3a030903271305m68c07a2fh767c3f5b2499aa@mail.gmail.com> if you prefix the sofia dial string with sip: you should be able to pass anything you want. sofia/internal/sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 2009/3/27 James H Thompson > Calls would be sent to the IP address after the '@' <%27@%27> in the > URI. > Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as the > user part of a SIP URI. > My example Invite URI is the way we are receiving traffic from some of the > major telecom carriers. > We would like be able to generate calls using the same formats. > > > ----- Original Message ----- *From:* Michael Jerris > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Friday, March 27, 2009 5:41 AM > *Subject:* Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls > > You seem to be confusing your standards, those 2 specs are about tel: uri's > not sip: uris. Sending a tel uri I am not sure we can do, where would we > send it to? > Mike > > On Mar 27, 2009, at 6:11 AM, James H Thompson wrote: > > I need to generate calls with Invite URIs in this format: > > INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 > SIP/2.0 > > Is there an easy way to do this? > > Thanks. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/acbdb383/attachment-0002.html From william.suffill at gmail.com Fri Mar 27 13:04:51 2009 From: william.suffill at gmail.com (William Suffill) Date: Fri, 27 Mar 2009 16:04:51 -0400 Subject: [Freeswitch-users] Contacting Callie In-Reply-To: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> References: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> Message-ID: <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> Good question. Last I talked to Brian about this (new prompts for upcoming new release) all the prompts are done by http://www.gmvoices.com/. I don't know anything more about the process to get recordings done or if there is any preferred process if they are from users of FreeSwitch but be curious to find out. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/8ac119f0/attachment-0002.html From anthony.minessale at gmail.com Fri Mar 27 13:06:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Mar 2009 15:06:44 -0500 Subject: [Freeswitch-users] Contacting Callie In-Reply-To: <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> References: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> Message-ID: <191c3a030903271306x37a851d1mcd90295f1837a574@mail.gmail.com> you can get the through us, just contact brian 2009/3/27 William Suffill > Good question. Last I talked to Brian about this (new prompts for upcoming > new release) all the prompts are done by http://www.gmvoices.com/. I don't > know anything more about the process to get recordings done or if there is > any preferred process if they are from users of FreeSwitch but be curious to > find out. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/be48521d/attachment-0002.html From brian at freeswitch.org Fri Mar 27 13:11:19 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 15:11:19 -0500 Subject: [Freeswitch-users] Contacting Callie In-Reply-To: <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> References: <2d9149cd0903271248v27fac1fel2ad16c1c35ff0666@mail.gmail.com> <6b65470d0903271304v4c9af3fam923281626b4b745e@mail.gmail.com> Message-ID: <1B17EFD7-AB3D-44B5-BDE4-9C6B82B8BFA1@freeswitch.org> Well I do get a discount if we batch them. I'm taking donations for this order brian at freeswitch.org is my paypal we are sending the order out monday but I only have a handful of stuff to record this go around. http://jira.freeswitch.org/browse/FSSCRIPTS-15 William, Thanks for your donation to help pay for it. ;) /b On Mar 27, 2009, at 3:04 PM, William Suffill wrote: > Good question. Last I talked to Brian about this (new prompts for > upcoming new release) all the prompts are done by http://www.gmvoices.com/ > . I don't know anything more about the process to get recordings > done or if there is any preferred process if they are from users of > FreeSwitch but be curious to find out. > > -- W Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/e792f639/attachment-0002.html From freeswitch at servercorps.com Fri Mar 27 13:14:42 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Fri, 27 Mar 2009 15:14:42 -0500 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> Message-ID: <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> Also, moving the list to Google Groups would allow email OR threaded views, and personally I like them better than nabble. anm_ On Thu, Mar 26, 2009 at 11:06 AM, Michael Jerris wrote: > http://n2.nabble.com/freeswitch-users-f2379917.html > > Mike > > > On Mar 26, 2009, at 12:01 PM, Tim Ringenbach wrote: > >> Is there nothing out there that integrates a forum with a mailing >> list? >> It seems like one could display the mailing list archives exactly >> like a >> forum, and allow users to register to the forum and post (appearing to >> the mailing list as username at forumurl.org) in such a way that they >> don't >> have to realize it's a mailing list. >> >> Anthony Minessale wrote: >>> The guy started a forum almost a month ago and as you can see nobody >>> knows the url and it has no posts. >>> >>> http://freeswitch411.info/forum/ >>> >>> This is one of the problems I was worried about when endorsing a >>> forum. >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rupa at rupa.com Fri Mar 27 13:22:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 27 Mar 2009 15:22:23 -0500 Subject: [Freeswitch-users] High CPU load but only few sessions In-Reply-To: <49CCEC15.8010500@ewetel.de> References: <49CB8D3D.7050202@ewetel.de> <3DA0B21A-33E6-49A0-905E-EBE20BB6E637@avgs.ca> <49CB98E1.8080705@ewetel.de> <49CB9E0C.4030300@ewetel.de> <49CCEC15.8010500@ewetel.de> Message-ID: kill -9 on a thread will kill the process which kills freeswitch. On Fri, Mar 27, 2009 at 10:09 AM, Helmut Kuper wrote: > Hello, > > today I killed that special thread via "kill -9" a simple kill didn't > helped. Unfortunately this led to a normal shutdown of FS although I > killed not the parent process. :( > > After restart of FS the server has a normal load again. > > regards and a nice weekend > Helmut > > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/8ab9bb89/attachment-0002.html From jht at lj.net Fri Mar 27 13:43:42 2009 From: jht at lj.net (James H Thompson) Date: Fri, 27 Mar 2009 10:43:42 -1000 Subject: [Freeswitch-users] IRC is not for all References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com><3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> Message-ID: <091e01c9af1c$b7a44130$0aa8a8c0@jhtp28> The freeswitch user mailing list is also on: http://news.gmane.org/gmane.comp.telephony.freeswitch.user There are several forums packages that allow feeding in a mailing list, although not many people seem to do it. Google Groups and Yahoo Groups are also alternatives. I've been considering mirroring some of the major voip mailing lists on voip-info.org into a forum of somekind. If this would be of interest let me know. Jim ----- Original Message ----- From: "Addison Martin" To: Sent: Friday, March 27, 2009 10:14 AM Subject: Re: [Freeswitch-users] IRC is not for all > Also, moving the list to Google Groups would allow email OR threaded > views, and personally I like them better than nabble. > > anm_ > > > On Thu, Mar 26, 2009 at 11:06 AM, Michael Jerris wrote: >> http://n2.nabble.com/freeswitch-users-f2379917.html >> >> Mike >> >> >> On Mar 26, 2009, at 12:01 PM, Tim Ringenbach wrote: >> >>> Is there nothing out there that integrates a forum with a mailing >>> list? >>> It seems like one could display the mailing list archives exactly >>> like a >>> forum, and allow users to register to the forum and post (appearing to >>> the mailing list as username at forumurl.org) in such a way that they >>> don't >>> have to realize it's a mailing list. >>> >>> Anthony Minessale wrote: >>>> The guy started a forum almost a month ago and as you can see nobody >>>> knows the url and it has no posts. >>>> >>>> http://freeswitch411.info/forum/ >>>> >>>> This is one of the problems I was worried about when endorsing a >>>> forum. >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jason at jasonjgw.net Fri Mar 27 18:42:55 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 28 Mar 2009 12:42:55 +1100 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <091e01c9af1c$b7a44130$0aa8a8c0@jhtp28> References: <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> <091e01c9af1c$b7a44130$0aa8a8c0@jhtp28> Message-ID: <20090328014255.GA10819@jdc.jasonjgw.net> James H Thompson wrote: > I've been considering mirroring some of the major voip mailing lists on > voip-info.org > into a forum of somekind. Have a look at http://www.gmane.org/ and note that you can post via NNTP or via the WEb. This mailing list is subscribed to gmane. From jason at jasonjgw.net Fri Mar 27 18:47:18 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 28 Mar 2009 12:47:18 +1100 Subject: [Freeswitch-users] IRC is not for all In-Reply-To: <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> References: <3181A30B8C35AB4AA8577B78DDF4613804A6913F@nickel.mettonigroup.com> <3181A30B8C35AB4AA8577B78DDF4613804B2AC88@nickel.mettonigroup.com> <191c3a030903260756i729e6d37l87ffb3175f8eb4e0@mail.gmail.com> <49CBA6CB.4030005@asteriasgi.com> <92e7d2090903271314r3727a66w96cd5e3d540ca470@mail.gmail.com> Message-ID: <20090328014718.GA10860@jdc.jasonjgw.net> Addison Martin wrote: > Also, moving the list to Google Groups would allow email OR threaded > views, and personally I like them better than nabble. http://dir.gmane.org/gmane.comp.telephony.freeswitch.user Would any of those views suffice? From dujinfang at gmail.com Fri Mar 27 19:22:23 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 28 Mar 2009 10:22:23 +0800 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <191c3a030903271204s73b96a92k77780034bc7b42be@mail.gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903260759k230ef4d3u1d8de1a36b2f0f8b@mail.gmail.com> <0DA25512-A208-4A92-8AB8-5DB463A860DB@gmail.com> <191c3a030903271204s73b96a92k77780034bc7b42be@mail.gmail.com> Message-ID: <0D693D25-7BC3-4D22-8D05-6103BADEB31A@gmail.com> Thanks. I was thinking created a new server with Ubuntu intrepid, seems I'd like back to Hardy. Even on hardy the default libtool is version 2. replace to libtool 1 should be easy as I mentioned before. On Mar 28, 2009, at 3:04 AM, Anthony Minessale wrote: > 1) There is an incompatibility on the fake ncurses wrapper that > causes an instant seg fault unless you install the real ncurses. On ubuntu it's libncurses5-dev, use for simple's sake. > > 2) The bleeding edge GCC builds an openzap binary that crashes > instantly with no explanation in the core file from a minimal -O2 > (that's just the one copmiler bug that we know about for sure, like > cock roaches, see one, there are probably 1000) we don't use openzap. Is the probably 1000 all in the openzap or anywhere else potentially?. > > 3) They upgraded to libtool 2.0 which builds binaries that will not > start. (easier said than done to upgrade ours too as we have to make > sure we work on *every* plarform and the upgrade to make it work > would break other operating systems we support) > Understand. > Bottom line, it's not their fault or anything but the choice to use > all brand new versions of everything under the sun is not a good > idea for your server, it's great that we have bleeding edge stuff or > we would not have anyone to test stuff, we have a similar group of > people always running SVN trunk of the day. But it's hard to > stabalize code when both your code and the OS may be unstable at the > same time. > > There is a reason they call it bleeding vs stable, which one would > you rather be if you were in the hospital. =D > As you mentioned. It's not their fault. ppl want to live on the edge just need to install multi-versions of gcc(or other tools). Like the Linux kernel, to compile from source, gcc-3 was recommended for a long time. Don't know if it's still the case recently. > > > 2009/3/27 dujinfang > > On Mar 26, 2009, at 10:59 PM, Anthony Minessale wrote: >> We do not support ubuntu interpid, it has at least 3 known fatal >> issues not experienced by all but nonetheless enough to make us >> unwilling to support it. > > I use Ubuntu gutsy in production and interipid in test env. It > works well. Can you briefly explain the 3 fatal issues Anthony? It > will help me know potential risks. > >> >> >> It's "use at your own risk" or use the stable branch "hardy" for >> any support. >> >> >> >> On Thu, Mar 26, 2009 at 4:45 AM, Trevor Hammonds > > wrote: >> Has there been any progress getting FreeSWITCH to build on Ubuntu >> Intrepid without downgrading libtool? >> > > I successfully built FS on intrepid. I simply done this by changing > the apt-source to Hardy and installed libtool. Obviously I changed > the apt-source back to intrepid after I installed libtool. > > And, another approach. Install libtool from source should be as easy > as configure && make && make install. I done this on a new CentOS4 > because the default yum install of libtool on CentOS4 is old than FS > required. > > >> Thanks! >> >> Sincerely, >> Trevor Hammonds >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> 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:213-799-1400 >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/9ab6bfca/attachment-0002.html From dujinfang at gmail.com Fri Mar 27 19:39:25 2009 From: dujinfang at gmail.com (dujinfang) Date: Sat, 28 Mar 2009 10:39:25 +0800 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <191c3a030903271305m68c07a2fh767c3f5b2499aa@mail.gmail.com> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> <191c3a030903271305m68c07a2fh767c3f5b2499aa@mail.gmail.com> Message-ID: <7F119E06-EC37-41EF-9919-45D2B9358880@gmail.com> On Mar 28, 2009, at 4:05 AM, Anthony Minessale wrote: > if you prefix the sofia dial string with sip: you should be able to > pass anything you want. > > sofia/internal/sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 > Is that similar as this? got it from wiki: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#From_the_Dialplan > 2009/3/27 James H Thompson > Calls would be sent to the IP address after the '@' in the URI. > Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as > the user part of a SIP URI. > My example Invite URI is the way we are receiving traffic from some > of the major telecom carriers. > We would like be able to generate calls using the same formats. > > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Friday, March 27, 2009 5:41 AM > Subject: Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls > > You seem to be confusing your standards, those 2 specs are about > tel: uri's not sip: uris. Sending a tel uri I am not sure we can > do, where would we send it to? > > Mike > > On Mar 27, 2009, at 6:11 AM, James H Thompson wrote: > >> I need to generate calls with Invite URIs in this format: >> >> INVITE sip:9085551212;npdi=yes;rn=9083820000 at 204.123.123.123:5060 >> SIP/2.0 >> >> Is there an easy way to do this? >> >> Thanks. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/4de5879c/attachment-0002.html From brian at freeswitch.org Fri Mar 27 19:55:41 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Mar 2009 21:55:41 -0500 Subject: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls In-Reply-To: <7F119E06-EC37-41EF-9919-45D2B9358880@gmail.com> References: <081301c9aec4$6f5208c0$0aa8a8c0@jhtp28> <08cb01c9af15$90ec2960$0aa8a8c0@jhtp28> <191c3a030903271305m68c07a2fh767c3f5b2499aa@mail.gmail.com> <7F119E06-EC37-41EF-9919-45D2B9358880@gmail.com> Message-ID: no sofia/profile/sip:blah at blah sip: makes sofia take it as is. /b On Mar 27, 2009, at 9:39 PM, dujinfang wrote: > > Is that similar as this? > > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090327/eeb3b0fb/attachment-0002.html From hads at nice.net.nz Fri Mar 27 21:23:47 2009 From: hads at nice.net.nz (Hadley Rich) Date: Sat, 28 Mar 2009 17:23:47 +1300 Subject: [Freeswitch-users] Building on Ubuntu Intrepid In-Reply-To: <0D693D25-7BC3-4D22-8D05-6103BADEB31A@gmail.com> References: <711825c70903260245j2c80487fne337ba50345b2ef2@mail.gmail.com> <191c3a030903271204s73b96a92k77780034bc7b42be@mail.gmail.com> <0D693D25-7BC3-4D22-8D05-6103BADEB31A@gmail.com> Message-ID: <200903281723.47626.hads@nice.net.nz> On Sat, 28 Mar 2009 15:22:23 dujinfang wrote: > Even on hardy the default libtool is version 2. replace to libtool 1 > should be easy as I mentioned before. The libtool on Hardy is 1.5.26 hads -- http://nicegear.co.nz VoIP, DVB and other Linux compatible hardware. From moizchinoy at gmail.com Sat Mar 28 02:09:02 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Sat, 28 Mar 2009 13:09:02 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <1B117EFC18604C6E82663FDAEB3342A8@quos> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> Message-ID: <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> I am trying it on windows. Where to get gnutls-devel for windows? On Fri, Mar 27, 2009 at 6:20 AM, Andrea wrote: > try to install ?gnutls-devel (name on centos) before to compile. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. From dave at 3c.co.uk Sat Mar 28 06:44:49 2009 From: dave at 3c.co.uk (David Knell) Date: Sat, 28 Mar 2009 07:44:49 -0600 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49C0F9EA.3000200@freeswitch.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BF942A.3030305@coppice.org> <49BFBBFD.1050308@3c.co.uk> <49BFC394.6070806@coppice.org> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> Message-ID: <49CE29D1.3000603@3c.co.uk> Raymond Chandler wrote: > What's interesting to me is.... everyone on this thread except you has > said that in real-world scenarios, they need the EC for reliability. > One of which, does signal processing programming professionally. It > seems to me that if you "build a better mouse trap" you must know what's > involved in making it work properly. I'm not sure what your background > really is, but you'd be hard pressed to match up to Steve's reputation > and/or experience. > Public willy-waving is undignified but, in brief, I've built and sold IVRs since 1997, wrote a CAPI-based soft IVR in 1999 (which required software for, inter alia, DTMF detection), developed a software fax modem (V.29, V.27ter, T.30, etc.) which I sold to a CTI card vendor and so on. I've collected some data, of which it is commonly said that the plural of anecdote - which is what we've had so far - is not. The IVR collects a 16 digit DTMF string, terminated by #. TDM->IP conversion was performed by an Asterisk box with an el-cheapo quad E1 card (no EC) for half the calls, and an AS5400 (with EC) for the other half. The proportion of entries missing one or more digit was 3.1% (Asterisk) and 3.3% (AS5400); this is not a statistically significant difference given the sample size. The reason for looking at this criterion is (a) that it's easy to measure, and (b) the most likely way that a DTMF detector will fail in the presence of excess noise, which includes echo, would be to miss a digit. This error rate is the sum of human error + detector error, and I've no measurements to show how this might be split; I would expect it's almost all human. Note that this is a digit error rate of about 1 in 500. This is, of course, only data from one site, but it's a start; it's only by collecting data such as this that one can understand how well one's mouse trap works and whether it needs improvement or not. > That said, it might be a good idea to just agree to disagree as this is > starting to sound like the faxing over IP talks I hear a lot. (i.e. > "faxing over g.711u with no t.38 works fine for me") Where it might work > for some people by some mysterious phenomena, it won't work for the > general public. And telling people that they don't need EC, where so > many have already said that they obviously do, is just as irresponsible, > IMHO, as you claiming Steve was for telling them that they don't need it. > That's a simplification. Simple IVR (record, replay, collect DTMF) probably doesn't need EC; if you're trying to do ASR with barge-in, bridge callers to other callers or operators, etc., then you probably do. I am interested that the recommended solution is 'buy Sangoma' - expensive and proprietary - when Oslec, a FOSS echo cancellers which, by all accounts, works extremely well, is out there and has been for some time. --Dave From mike at jerris.com Sat Mar 28 09:03:08 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 28 Mar 2009 12:03:08 -0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> Message-ID: <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> please see my previous response in this thread. MIke On Mar 28, 2009, at 5:09 AM, Moiz Chinoy wrote: > I am trying it on windows. > Where to get gnutls-devel for windows? > From mszlazak at aol.com Sat Mar 28 10:58:20 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Sat, 28 Mar 2009 13:58:20 -0400 Subject: [Freeswitch-users] switch_ivr_originate() Parse Error! Message-ID: <8CB7DE5EA49B71E-A84-2D59@WEBMAIL-MY06.sysops.aol.com> I'm getting a parsing error which seems to be coming from the space in "Extension 1000". If this is "normal" then what's the best way to deal with spaces in caller id names? This is how the call was originated: ??? ??? ??? 2009-03-28 10:50:56 [ERR] switch_ivr_originate.c:976 switch_ivr_originate() Parse Error! 2009-03-28 10:50:56 [DEBUG] switch_ivr_originate.c:2081 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2009-03-28 10:50:56 [DEBUG] mod_commands.c:2213 sch_api_callback() Command originate({id_name=Extension 1000,id_number=1000}sofia/internal/1000%10.0.0.3 GINO_ANS): -ERR DESTINATION_OUT_OF_ORDER Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/ff950b85/attachment-0002.html From brian at freeswitch.org Sat Mar 28 11:13:44 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Mar 2009 13:13:44 -0500 Subject: [Freeswitch-users] switch_ivr_originate() Parse Error! In-Reply-To: <8CB7DE5EA49B71E-A84-2D59@WEBMAIL-MY06.sysops.aol.com> References: <8CB7DE5EA49B71E-A84-2D59@WEBMAIL-MY06.sysops.aol.com> Message-ID: <303F2923-BD29-4924-BE54-BBAFF47C139F@freeswitch.org> quote it! Single quote '${blah}' /b On Mar 28, 2009, at 12:58 PM, mszlazak at aol.com wrote: > I'm getting a parsing error which seems to be coming from the space > in "Extension 1000". If this is "normal" then what's the best way to > deal with spaces in caller id names? Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/3a272c5b/attachment-0002.html From bipin at xbipin.com Sat Mar 28 12:21:37 2009 From: bipin at xbipin.com (xbipin) Date: Sat, 28 Mar 2009 12:21:37 -0700 (PDT) Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> Message-ID: <22760426.post@talk.nabble.com> hi, im trying to do the same, use FS as a plain and simple SBC but cant figure out how to do so -- View this message in context: http://www.nabble.com/freeswitch-as-a-session-border-controller-tp16755838p22760426.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Sat Mar 28 15:35:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 28 Mar 2009 17:35:35 -0500 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49CE29D1.3000603@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> Message-ID: <191c3a030903281535o6f3d09dbye20cf947cd250d80@mail.gmail.com> Well, >From my experience, an AS5300 thinks nearly *anthing* is dtmf. The one I have as a PSTN onramp to our conference bridge drives me bonkers with false positives since every key on the pad means something in our default conference. Jim Dixon designed the el-cheapo (tormenta 2) to put as many resources on the host as possible, that was the goal behind the initiative. The driver for this card was the template on which the entire Zaptel (now Dahdi) was based. What's interesting is that early zapata library (a user space abstraction lib) was completely consumed by chan_zap and from there may of the featues were gravitated towards the linux kernel then eventually into the hardware as new cards were developed. The typical reason for this kind of evolution is customers. When there was no TDM to be had at all, el-cheapo and software was the bees knees. As they started getting more greedy and anxious for higher quality, they started asking for improvements that lead to more stuff onboard in the new cards. I am guessing this will continue until the cards are too expensive and we will go full circle back to all-on-host just in time for the 16 core CPU box being standard issue. It all depends on the strategy employed, if you want to use a tor2 and oslec, (a software echo canceller that, in fact has none other than Steve Underwood from this thread as a collaberator) then do it. If it ain't broke, don't fix it.....If you prefer to have hardware EC, then buy a card that supports it. ------ see http://www.rowetel.com/ucasterisk/oslec.html Background and Credits Oslec started life as a prototype echo canceller and G168 test framework from Steve Underwood's Spandsp library. Steve wrote much of the DSP code used in Asterisk, and the Zaptel echo cancellation code is heavily based on his work. ------ Bottom line: There is no real correct answer because it depends on what your goals are and what you personally prefer. I personally have used both, I am annoyed with hardware EC because it breaks software dtmf but now the sangoma drivers have hardware dtmf to use together with hardware EC so that solves the problem. I would prefer not to take any sides in this debate since everyone on this thread has contributed greatly to our project and I respect them all. I would, however, like to ask that maybe we can channel all of this intelligence into some common goal and do somethng great rather than spend our energy doing the techno-geek version of m&m freestyle rapping. On Sat, Mar 28, 2009 at 8:44 AM, David Knell wrote: > Raymond Chandler wrote: > > What's interesting to me is.... everyone on this thread except you has > > said that in real-world scenarios, they need the EC for reliability. > > One of which, does signal processing programming professionally. It > > seems to me that if you "build a better mouse trap" you must know what's > > involved in making it work properly. I'm not sure what your background > > really is, but you'd be hard pressed to match up to Steve's reputation > > and/or experience. > > > Public willy-waving is undignified but, in brief, I've built and sold > IVRs since > 1997, wrote a CAPI-based soft IVR in 1999 (which required software for, > inter > alia, DTMF detection), developed a software fax modem (V.29, V.27ter, T.30, > etc.) which I sold to a CTI card vendor and so on. > > I've collected some data, of which it is commonly said that the plural > of anecdote - > which is what we've had so far - is not. The IVR collects a 16 digit > DTMF string, > terminated by #. TDM->IP conversion was performed by an Asterisk box with > an el-cheapo quad E1 card (no EC) for half the calls, and an AS5400 > (with EC) > for the other half. > > The proportion of entries missing one or more digit was 3.1% (Asterisk) > and 3.3% > (AS5400); this is not a statistically significant difference given the > sample size. > The reason for looking at this criterion is (a) that it's easy to > measure, and (b) the > most likely way that a DTMF detector will fail in the presence of excess > noise, > which includes echo, would be to miss a digit. This error rate is the > sum of > human error + detector error, and I've no measurements to show how this > might > be split; I would expect it's almost all human. Note that this is a > digit error rate of > about 1 in 500. > > This is, of course, only data from one site, but it's a start; it's only > by collecting > data such as this that one can understand how well one's mouse trap works > and whether it needs improvement or not. > > That said, it might be a good idea to just agree to disagree as this is > > starting to sound like the faxing over IP talks I hear a lot. (i.e. > > "faxing over g.711u with no t.38 works fine for me") Where it might work > > for some people by some mysterious phenomena, it won't work for the > > general public. And telling people that they don't need EC, where so > > many have already said that they obviously do, is just as irresponsible, > > IMHO, as you claiming Steve was for telling them that they don't need it. > > > That's a simplification. Simple IVR (record, replay, collect DTMF) > probably > doesn't need EC; if you're trying to do ASR with barge-in, bridge callers > to > other callers or operators, etc., then you probably do. > > I am interested that the recommended solution is 'buy Sangoma' - expensive > and proprietary - when Oslec, a FOSS echo cancellers which, by all > accounts, > works extremely well, is out there and has been for some time. > > --Dave > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090328/c79e865f/attachment-0002.html From saeedahmad1981 at gmail.com Sat Mar 28 16:49:53 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Sun, 29 Mar 2009 00:49:53 +0100 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <22760426.post@talk.nabble.com> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> Message-ID: <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> http://wiki.freeswitch.org/wiki/SBC_Setup written by rod. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of xbipin Sent: Saturday, March 28, 2009 8:22 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a session border controller hi, im trying to do the same, use FS as a plain and simple SBC but cant figure out how to do so -- View this message in context: http://www.nabble.com/freeswitch-as-a-session-border-controller-tp16755838p2 2760426.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From f.koliqi at gmail.com Sat Mar 28 22:30:36 2009 From: f.koliqi at gmail.com (Fadil Berisha) Date: Sun, 29 Mar 2009 01:30:36 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49CE29D1.3000603@3c.co.uk> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> Message-ID: <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> > That's a simplification. Simple IVR (record, replay, collect DTMF) > probably > doesn't need EC; > Dave Collect DTMF does not need EC. I take out your word "probably" because no need for any dilemma. Interaction between DTMF detector an EC *when EC exist * is different question and deserve separate thread. Although I am confirming your statement,** I can not say "I am voting for you", simple because this is not political forum to express believing to one or other leader or authority. With respect koliqi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090329/689b231b/attachment-0002.html From steveu at coppice.org Sun Mar 29 01:32:56 2009 From: steveu at coppice.org (Steve Underwood) Date: Sun, 29 Mar 2009 16:32:56 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49BFE1F3.2030207@3c.co.uk> <49BFEBCB.9020708@coppice.org> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> Message-ID: <49CF3238.9040503@coppice.org> Fadil Berisha wrote: > > That's a simplification. Simple IVR (record, replay, collect > DTMF) probably > doesn't need EC; > > > > Dave > > Collect DTMF does not need EC. I take out your word "probably" > because no need for any dilemma. Interaction between DTMF detector an > EC *when EC exist* is different question and deserve separate thread. > Although I am confirming your statement,// I can not say "I am voting > for you", simple because this is not political forum to express > believing to one or other leader or authority. Receiving DTMF reliably needs a signal to noise ratio of about 10dB if the noise is Gaussian. The statistics of voice mean you need the DTMF to be more like 15dB above voice. Most hybrids are only required to have a return loss of better than 12dB. They can be *much* better, but don't count on it, especially at the phone's hybrid. You have multiple hybrids in the path (usually 2 or 4). Let's take the better case with only 2 sitting between the outgoing exchange card and the far end phone. Put 10km of copper between the exchange the phone (typical copper planning limit) and you probably have 15dB of attenuation on the line. Now your DTMF is actually below the level of the voice prompt for much of the time. Think that will work with an echo canceller? Sure you can get reliable DTMF detection on 70%-80% of call paths with no echo cancellation, but if you want reliability with close to 100% of phone lines, you need echo cancellation to remove the voice prompt from the signal received at the IVR. Dialogic, NMS, and the others didn't put EC on their cards for nothing. The only reason the normal connection of a phone to a line card gets reliable detection of the first dialed digit in the presence of a dialing tone is that the DTMF detector heavily filters that dialing tone. Forget the political forum crap. If you want to refute what I just said, try to back up your argument with some actual engineering. Regards, Steve From bipin at xbipin.com Sat Mar 28 22:50:52 2009 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 29 Mar 2009 09:50:52 +0400 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> Message-ID: <49CF0C3C.8070508@xbipin.com> hi, what are the other steps if done on windows platform with FS as SBC and Voipswitch as the server. secondly how to make the clients registering to voipswitch go through FS in which case FS should simply accept all registrations but it should do a forward registration to voipswitch and if voipswitch rejects then FS should also reject. basically i need FS as a SBC with topology hiding. Regards, Bipin www.xbipin.com +971-55-9270058 -------- Original Message -------- Subject: Re: [Freeswitch-users] freeswitch as a session border controller From: Saeed Ahmed To: freeswitch-users at lists.freeswitch.org Date: Sunday, March 29, 2009 3:49:53 AM > http://wiki.freeswitch.org/wiki/SBC_Setup written by rod. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of xbipin > Sent: Saturday, March 28, 2009 8:22 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] freeswitch as a session border controller > > > hi, > > im trying to do the same, use FS as a plain and simple SBC but cant figure > out how to do so From bipin at xbipin.com Sat Mar 28 23:03:34 2009 From: bipin at xbipin.com (xbipin) Date: Sat, 28 Mar 2009 23:03:34 -0700 (PDT) Subject: [Freeswitch-users] how to do upper registration Message-ID: <22743580.post@talk.nabble.com> i have been trying to do this but seem to have been lost, actually i want to use freeswitch as a session border controller so basically all the clients that try to register to FS will actually be authenticated voipswitch but i want FS to be in between and proxy everything and with any of its advanced options turned off. -- View this message in context: http://www.nabble.com/how-to-do-upper-registration-tp22743580p22743580.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From bipin at xbipin.com Sat Mar 28 23:07:02 2009 From: bipin at xbipin.com (xbipin) Date: Sat, 28 Mar 2009 23:07:02 -0700 (PDT) Subject: [Freeswitch-users] upper registration in FS? Message-ID: <22764757.post@talk.nabble.com> i have been trying to do this but seem to have been lost, actually i want to use freeswitch as a session border controller so basically all the clients that try to register to FS will actually be authenticated by voipswitch but i want FS to be in between and proxy everything and with any of its advanced options turned off so as to simply work as topology hiding, proxy, SBC only. -- View this message in context: http://www.nabble.com/upper-registration-in-FS--tp22764757p22764757.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From krice at suspicious.org Sun Mar 29 12:07:23 2009 From: krice at suspicious.org (Ken Rice) Date: Sun, 29 Mar 2009 14:07:23 -0500 Subject: [Freeswitch-users] how to do upper registration In-Reply-To: <22743580.post@talk.nabble.com> Message-ID: FreeSwitch is NOT a proxy... It is a B2BUA... You need to look at something like OpenSIPS/OpenSER for what you are trying to do. > From: xbipin > Reply-To: > Date: Sat, 28 Mar 2009 23:03:34 -0700 (PDT) > To: > Subject: [Freeswitch-users] how to do upper registration > > > i have been trying to do this but seem to have been lost, actually i want to > use freeswitch as a session border controller so basically all the clients > that try to register to FS will actually be authenticated voipswitch but i > want FS to be in between and proxy everything and with any of its advanced > options turned off. > -- > View this message in context: > http://www.nabble.com/how-to-do-upper-registration-tp22743580p22743580.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From grevenx at me.com Sun Mar 29 14:16:13 2009 From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Sun, 29 Mar 2009 23:16:13 +0200 Subject: [Freeswitch-users] how to do upper registration In-Reply-To: References: Message-ID: While it's not a proxy, FS CAN be a SBC as he mentions (though I'm not sure the feature he describes fits the bill for a SBC?). http://en.wikipedia.org/wiki/Back-to-back_user_agent http://en.wikipedia.org/wiki/Session_Border_Controller http://wiki.freeswitch.org/wiki/Specsheet Best regards, Even Andr? On 29. mars. 2009, at 21.07, Ken Rice wrote: > FreeSwitch is NOT a proxy... It is a B2BUA... You need to look at > something > like OpenSIPS/OpenSER for what you are trying to do. > > > > >> From: xbipin >> Reply-To: >> Date: Sat, 28 Mar 2009 23:03:34 -0700 (PDT) >> To: >> Subject: [Freeswitch-users] how to do upper registration >> >> >> i have been trying to do this but seem to have been lost, actually >> i want to >> use freeswitch as a session border controller so basically all the >> clients >> that try to register to FS will actually be authenticated >> voipswitch but i >> want FS to be in between and proxy everything and with any of its >> advanced >> options turned off. >> -- >> View this message in context: >> http://www.nabble.com/how-to-do-upper-registration-tp22743580p22743580.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dave.cabot at iconsoluciones.net Sun Mar 29 13:16:26 2009 From: dave.cabot at iconsoluciones.net (Dave Cabot) Date: Sun, 29 Mar 2009 16:16:26 -0400 Subject: [Freeswitch-users] mod_skypiax Message-ID: <1238357786.7199.29.camel@dave-laptop> Hi folks. I've been playing with mod_skypiax. 1) it seems that the documentation web pages are out of date. http://jira.freeswitch.org/ rejects connections. I'm running FreeSWITCH Version 1.0.trunk (12826). I'm using Xvfb and skype 2.0.0.72-1_i386.deb. I can make a call and sound is great, but then when the call ends, so does skype. Attached is what gdb puked: root at freeswitch-1:/usr/local/freeswitch/conf# gdb /usr/bin/skype 20250 GNU gdb 6.8-debian Copyright (C) 2008 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "i486-linux-gnu"... (no debugging symbols found) Attaching to program: /usr/bin/skype, process 20250 Reading symbols from /usr/lib/libasound.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libasound.so.2 Reading symbols from /usr/lib/libXv.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXv.so.1 Reading symbols from /usr/lib/libXss.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXss.so.1 Reading symbols from /lib/tls/i686/cmov/librt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/librt.so.1 Reading symbols from /usr/lib/libQtDBus.so.4... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtDBus.so.4 Reading symbols from /usr/lib/libQtGui.so.4...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtGui.so.4 Reading symbols from /usr/lib/libQtNetwork.so.4... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtNetwork.so.4 Reading symbols from /usr/lib/libQtCore.so.4...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtCore.so.4 Reading symbols from /lib/tls/i686/cmov/libpthread.so.0... (no debugging symbols found)...done. [Thread debugging using libthread_db enabled] [New Thread 0xb6ef16c0 (LWP 20250)] [New Thread 0xb5708b90 (LWP 20261)] [New Thread 0xb6116b90 (LWP 20259)] [New Thread 0xb6197b90 (LWP 20256)] [New Thread 0xb6218b90 (LWP 20255)] [New Thread 0xb6a19b90 (LWP 20254)] [New Thread 0xb6a9ab90 (LWP 20253)] [New Thread 0xb6db3b90 (LWP 20251)] Loaded symbols for /lib/tls/i686/cmov/libpthread.so.0 Reading symbols from /usr/lib/libstdc++.so.6...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libstdc++.so.6 Reading symbols from /lib/tls/i686/cmov/libm.so.6... (no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libm.so.6 Reading symbols from /lib/libgcc_s.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libgcc_s.so.1 Reading symbols from /lib/tls/i686/cmov/libc.so.6... (no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libc.so.6 Reading symbols from /usr/lib/libX11.so.6...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libX11.so.6 Reading symbols from /usr/lib/libXext.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXext.so.6 Reading symbols from /lib/tls/i686/cmov/libdl.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libdl.so.2 Reading symbols from /lib/ld-linux.so.2... (no debugging symbols found)...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/lib/libdbus-1.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libdbus-1.so.3 Reading symbols from /usr/lib/libQtXml.so.4... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtXml.so.4 Reading symbols from /usr/lib/libfontconfig.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libfontconfig.so.1 Reading symbols from /usr/lib/libz.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libz.so.1 Reading symbols from /usr/lib/libgthread-2.0.so.0...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libgthread-2.0.so.0 Reading symbols from /usr/lib/libglib-2.0.so.0... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libglib-2.0.so.0 Reading symbols from /usr/lib/libaudio.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libaudio.so.2 Reading symbols from /usr/lib/libXt.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXt.so.6 Reading symbols from /usr/lib/libpng12.so.0...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libpng12.so.0 Reading symbols from /usr/lib/libSM.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libSM.so.6 Reading symbols from /usr/lib/libICE.so.6...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libICE.so.6 Reading symbols from /usr/lib/libXi.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXi.so.6 Reading symbols from /usr/lib/libXrender.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXrender.so.1 Reading symbols from /usr/lib/libXrandr.so.2... ---Type to continue, or q to quit--- (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXrandr.so.2 Reading symbols from /usr/lib/libXfixes.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXfixes.so.3 Reading symbols from /usr/lib/libXcursor.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXcursor.so.1 Reading symbols from /usr/lib/libXinerama.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXinerama.so.1 Reading symbols from /usr/lib/libfreetype.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libfreetype.so.6 Reading symbols from /usr/lib/libxcb-xlib.so.0...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libxcb-xlib.so.0 Reading symbols from /usr/lib/libxcb.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libxcb.so.1 Reading symbols from /usr/lib/libXau.so.6...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libXau.so.6 Reading symbols from /usr/lib/libexpat.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libexpat.so.1 Reading symbols from /usr/lib/libpcre.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libpcre.so.3 Reading symbols from /usr/lib/libXdmcp.so.6... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libXdmcp.so.6 Reading symbols from /usr/lib/gconv/UTF-16.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/gconv/UTF-16.so Reading symbols from /usr/lib/qt4/plugins/imageformats/libqgif.so... (no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqgif.so Reading symbols from /usr/lib/qt4/plugins/imageformats/libqjpeg.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqjpeg.so Reading symbols from /usr/lib/libjpeg.so.62... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libjpeg.so.62 Reading symbols from /usr/lib/qt4/plugins/imageformats/libqmng.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqmng.so Reading symbols from /usr/lib/libmng.so.1... (no debugging symbols found)...done. Loaded symbols for /usr/lib/libmng.so.1 Reading symbols from /usr/lib/liblcms.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/liblcms.so.1 Reading symbols from /usr/lib/qt4/plugins/imageformats/libqsvg.so... (no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqsvg.so Reading symbols from /usr/lib/libQtSvg.so.4...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libQtSvg.so.4 Reading symbols from /usr/lib/qt4/plugins/imageformats/libqtiff.so... (no debugging symbols found)...done. Loaded symbols for /usr/lib/qt4/plugins/imageformats/libqtiff.so Reading symbols from /usr/lib/i686/cmov/libssl.so.0.9.8...(no debugging symbols found)...done. Loaded symbols for /usr/lib/i686/cmov/libssl.so.0.9.8 Reading symbols from /usr/lib/i686/cmov/libcrypto.so.0.9.8... (no debugging symbols found)...done. Loaded symbols for /usr/lib/i686/cmov/libcrypto.so.0.9.8 Reading symbols from /usr/lib/libresolv.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libresolv.so Reading symbols from /lib/tls/i686/cmov/libnss_files.so.2... (no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libnss_files.so.2 Reading symbols from /lib/tls/i686/cmov/libnss_dns.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/tls/i686/cmov/libnss_dns.so.2 (no debugging symbols found) 0xb7fdc410 in __kernel_vsyscall () (gdb) c Continuing. [New Thread 0xb6093b90 (LWP 20278)] [New Thread 0xb4eeab90 (LWP 20279)] [New Thread 0xb46e9b90 (LWP 20280)] RtApiAlsa: callback thread error (RtApiAlsa: audio write error for device (Default device (default)): Exec format error.) ... closing stream. [Thread 0xb4eeab90 (LWP 20279) exited] Skype Xv: Xv ports available: 0 Skype XShm: XShm support enabled [New Thread 0xb3e9cb90 (LWP 20283)] RtApiAlsa: callback thread error (RtApiAlsa: audio read error for device (Default device (default)): Unknown error 405.) ... closing stream. [Thread 0xb46e9b90 (LWP 20280) exited] [Thread 0xb3e9cb90 (LWP 20283) exited] [New Thread 0xb3d3db90 (LWP 20288)] Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 0xb6197b90 (LWP 20256)] 0xb738c4ac in free () from /lib/tls/i686/cmov/libc.so.6 (gdb) Continuing. Program received signal SIGABRT, Aborted. 0xb7fdc410 in __kernel_vsyscall () (gdb) bt #0 0xb7fdc410 in __kernel_vsyscall () #1 0xb7348085 in raise () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7349a01 in abort () from /lib/tls/i686/cmov/libc.so.6 #3 0x082f372a in ?? () #4 #5 0xb738c4ac in free () from /lib/tls/i686/cmov/libc.so.6 #6 0x08808972 in ?? () Backtrace stopped: previous frame inner to this frame (corrupt stack?) (gdb) quit The program is running. Quit anyway (and detach it)? (y or n) y Detaching from program: /usr/bin/skype, process 20250 root at freeswitch-1:/usr/local/freeswitch/conf# -bash: line 320: 20249 Done echo "blah blah" 20250 Aborted (core dumped) | DISPLAY=:101 /usr/bin/skype --pipelogin [1]+ Exit 134 ( echo "blah blah" | DISPLAY=:101 /usr/bin/skype --pipelogin ) Not only do I have to restart skype, but I also have to reload mod_skypiax. Obviously this has it's drawbacks. Has anyone seen this and have a solution tucked away? Thanks From can_man at gmx.de Sun Mar 29 15:09:21 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Mon, 30 Mar 2009 00:09:21 +0200 Subject: [Freeswitch-users] FS - MjSip no voice Message-ID: <20090329220921.227090@gmx.net> Hello everyone, I am trying to get FS working with the MjSip Java Sip-stack, the SipToSis source and the normal one. Everything works well within my own network and when using x-lite, but when it comes to making calls from MjSip to an outside FS server I don't hear any voice - seems to be a NAT problem or some kind of other MjSip problem. Registration works fine though and SIP messages get through ok, but non of the UDP RTP ones. Would be great if someone could advice me on how to do the setup correctly. The whole FS trace can be found here: http://pastebin.freeswitch.org/8029 The settings for MjSip are: "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090", "transport_protocols=udp tcp","from_url=", "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", "bin_rat=rat","bin_vic=vic" Thank you very much. Best wishes, Phil -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From brian at freeswitch.org Sun Mar 29 16:50:21 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 29 Mar 2009 18:50:21 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] mod_skypiax In-Reply-To: <1238357786.7199.29.camel@dave-laptop> References: <1238357786.7199.29.camel@dave-laptop> Message-ID: <5C65F009-BAC1-40EF-BD69-46547F267346@freeswitch.org> Please redirect that to jira. I have fixed it. /b On Mar 29, 2009, at 3:16 PM, Dave Cabot wrote: > 1) it seems that the documentation web pages are out of date. > http://jira.freeswitch.org/ rejects connections. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090329/6d709f04/attachment-0002.html From f.koliqi at gmail.com Sun Mar 29 19:20:20 2009 From: f.koliqi at gmail.com (Fadil Berisha) Date: Sun, 29 Mar 2009 22:20:20 -0400 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <49CF3238.9040503@coppice.org> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> <49CF3238.9040503@coppice.org> Message-ID: <5c7d82f20903291920od56993eu7a30a760e0ed4847@mail.gmail.com> > Sure you can get reliable DTMF detection on 70%-80% of call paths with > no echo cancellation, Fair enough. You say "sure you can get reliable DTMF detection on 70%-80% of call paths with no echo cancellation". OK, you forgot to mention that this is achievable with text-book algorithms and that exist advanced algorithms with reliability close to 100%. From my point of view , no need further arguing this issue and this thread for me is closed. With respect koliqi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090329/dac37785/attachment-0002.html From christian.bourke1 at gmail.com Sun Mar 29 17:55:14 2009 From: christian.bourke1 at gmail.com (Christian Bourke) Date: Mon, 30 Mar 2009 10:55:14 +1000 Subject: [Freeswitch-users] H323 supported devices Message-ID: <564d7da90903291755x1caf83e7k4c2e74a2761ba24c@mail.gmail.com> Hi, I would like to implement freeswitch as a H323 gateway. I would like to be able to receive calls from H323 devices such as Polycom V500 and Dlink DVC-1000. Does anyone know if this is possible? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/2f144d24/attachment-0002.html From krice at freeswitch.org Sun Mar 29 22:58:58 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 30 Mar 2009 00:58:58 -0500 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: <564d7da90903291755x1caf83e7k4c2e74a2761ba24c@mail.gmail.com> Message-ID: Yes its possible... Look at mod_opal From: Christian Bourke Reply-To: Date: Mon, 30 Mar 2009 10:55:14 +1000 To: Subject: [Freeswitch-users] H323 supported devices Hi, ? I would like to implement freeswitch as a H323 gateway. I would like to be able to receive calls from H323 devices such as Polycom V500 and Dlink DVC-1000. ? Does anyone know if this is possible? ? _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/97589ec2/attachment-0002.html From moizchinoy at gmail.com Mon Mar 30 00:56:46 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Mon, 30 Mar 2009 11:56:46 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> Message-ID: <29b888f80903300056t51828330s39bbf662fe2cd8c0@mail.gmail.com> Hi, If there is no build for the tls with freeswitch on windows, can you please guide how to test gtalk integration on windows. I have successfully compiled the tls with iksemel, there were couple of errors but I managed to compile it. Now, I am also getting the error libdingaling.c:1545 xmpp_connect() io error 2 7 What is this error about? There is a similar post on the mailing list regarding the above IO error but can't find the reply for it. On Sat, Mar 28, 2009 at 8:03 PM, Michael Jerris wrote: > please see my previous response in this thread. > > MIke > > On Mar 28, 2009, at 5:09 AM, Moiz Chinoy wrote: > >> I am trying it on windows. >> Where to get gnutls-devel for windows? >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. From christian.bourke1 at gmail.com Mon Mar 30 01:28:08 2009 From: christian.bourke1 at gmail.com (Christian Bourke) Date: Mon, 30 Mar 2009 18:28:08 +1000 Subject: [Freeswitch-users] Freeswitch Hunt Group Message-ID: <564d7da90903300128q70196f17r4834ec7ffb933fcd@mail.gmail.com> Hi, I would like to implement freeswitch as an H323 gatekeeper. I would like to know if its possible for incoming H323 calls to go into a hunt group, if no 'operators' are logged into the hunt group then the call will bounce to an auto attendant and the H323 caller will be played a short video message then freeswitch will disconnect the call. Does this sound possible with Freeswitch? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/6802b65d/attachment-0002.html From christian.bourke1 at gmail.com Mon Mar 30 01:30:49 2009 From: christian.bourke1 at gmail.com (Christian Bourke) Date: Mon, 30 Mar 2009 18:30:49 +1000 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: References: <564d7da90903291755x1caf83e7k4c2e74a2761ba24c@mail.gmail.com> Message-ID: <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> Thank you Ken. Do you know if mod_opal supports both H323 voice and video? 2009/3/30 Ken Rice > Yes its possible... Look at mod_opal > > > > ------------------------------ > *From: *Christian Bourke > *Reply-To: * > *Date: *Mon, 30 Mar 2009 10:55:14 +1000 > *To: * > *Subject: *[Freeswitch-users] H323 supported devices > > Hi, > > I would like to implement freeswitch as a H323 gateway. I would like to be > able to receive calls from H323 devices such as Polycom V500 and Dlink > DVC-1000. > > Does anyone know if this is possible? > > > ------------------------------ > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/8aee4f2c/attachment-0002.html From fdelawarde at wirelessmundi.com Mon Mar 30 01:34:10 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Mon, 30 Mar 2009 10:34:10 +0200 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues In-Reply-To: <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> References: <1238155504.4364.222.camel@localhost.localdomain> <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> Message-ID: <1238402050.4364.274.camel@localhost.localdomain> Thanks for your quick&clear answers. On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote: > If you are using on-hook agents, it will place as many outbound calls > as there are people waiting. > If you are using off-hook agents it will just connect the first free > agent. By "people waiting" you mean "calls in the queue" and not "agents waiting" right? > > - Is there an [easy] way (with some javascript or similar) to > "emulate" Asterisk's distribution strategies to agents (by > amount of time without calls, total number of answered calls, > round robing, ...) in this cases? > > > Easiest way would be to write a patch in C to mod_fifo it'self or > propose a bounty for features and see if you can get the change > approved by the developers. I'll give the patch a try once I have a bit more practice with FreeSWITCH (never even launched it yet) and its API. Any hints or "quick-start" guides for module writing? Fran?ois. From monemran at gmail.com Mon Mar 30 01:34:34 2009 From: monemran at gmail.com (M.Emran) Date: Mon, 30 Mar 2009 14:34:34 +0600 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> References: <564d7da90903291755x1caf83e7k4c2e74a2761ba24c@mail.gmail.com> <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> Message-ID: How is the performance for mod_opal ? is the performance same as SIP ? 2009/3/30 Christian Bourke > Thank you Ken. Do you know if mod_opal supports both H323 voice and video? > > 2009/3/30 Ken Rice > >> Yes its possible... Look at mod_opal >> >> >> >> ------------------------------ >> *From: *Christian Bourke >> *Reply-To: * >> *Date: *Mon, 30 Mar 2009 10:55:14 +1000 >> *To: * >> *Subject: *[Freeswitch-users] H323 supported devices >> >> Hi, >> >> I would like to implement freeswitch as a H323 gateway. I would like to be >> able to receive calls from H323 devices such as Polycom V500 and Dlink >> DVC-1000. >> >> Does anyone know if this is possible? >> >> >> ------------------------------ >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards ---------- M Emran Managing Director InSpiration Software Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/fadb1f3c/attachment-0002.html From krice at suspicious.org Mon Mar 30 03:40:13 2009 From: krice at suspicious.org (Ken Rice) Date: Mon, 30 Mar 2009 05:40:13 -0500 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> Message-ID: I?m not sure if the video support is complete but both freeswitch and opal support video... Now I also see another email you send asking about freeswitch being a gatekeeper... FreeSWITCH is just another h323 end point, it does not do all gatekeeper functions like keep up with what IP what h323 user is on... However, as on your other message it might be possible to do what you want... My suggestion would be to set it up and try... There needs to be more h323 testing anyway Ken From: Christian Bourke Reply-To: Date: Mon, 30 Mar 2009 18:30:49 +1000 To: Subject: Re: [Freeswitch-users] H323 supported devices Thank you Ken. Do you know if?mod_opal supports both H323 voice and video? 2009/3/30 Ken Rice > Yes its possible... Look at mod_opal > > > > > From: Christian Bourke > Reply-To: > Date: Mon, 30 Mar 2009 10:55:14 +1000 > To: > Subject: [Freeswitch-users] H323 supported devices > > > Hi, > ? > I would like to implement freeswitch as a H323 gateway. I would like to be > able to receive calls from H323 devices such as Polycom V500 and Dlink > DVC-1000. > ? > Does anyone know if this is possible? > ? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/441e43bc/attachment-0002.html From bipin at xbipin.com Sun Mar 29 22:28:11 2009 From: bipin at xbipin.com (xbipin) Date: Sun, 29 Mar 2009 22:28:11 -0700 (PDT) Subject: [Freeswitch-users] how to do upper registration In-Reply-To: References: <22743580.post@talk.nabble.com> Message-ID: <22776631.post@talk.nabble.com> so what i make of it is that FS cant do upper registration or lets say it wont proxy everything and wont use the registration details it receives from clients registering to it and use it to register clients to a different server for which its proxying, or mayb its not meant to do so but then i ask can FS be configured to accept all registrations coming to it without each userid and password being defined in the profiles or can FS be configured to simply accept all incomming calls without the need for clients to register to it in which case it will fullfill my needs partially. or one more thing is use the id and pass sent by client to be forwarded to my actual server and then its up to the server to authenticate it and connect the call or to reply with incorrect id and pass. this can be done only if FS could accept all registrations without it being defined in its profile and some how the dialplan was crafted to use the registration id and pass sent by client to be used as the id and pass for the SIP TRUNK or the server in general. -- View this message in context: http://www.nabble.com/how-to-do-upper-registration-tp22770442p22776631.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From bipin at xbipin.com Mon Mar 30 01:21:39 2009 From: bipin at xbipin.com (xbipin) Date: Mon, 30 Mar 2009 01:21:39 -0700 (PDT) Subject: [Freeswitch-users] how to do upper registration Message-ID: <22776631.post@talk.nabble.com> so what i make of it is that FS cant do upper registration or lets say it wont proxy everything and wont use the registration details it receives from clients registering to it and use it to register clients to a different server for which its proxying, or mayb its not meant to do so but then i ask can FS be configured to accept all registrations coming to it without each userid and password being defined in the profiles or can FS be configured to simply accept all incomming calls without the need for clients to register to it in which case it will fullfill my needs partially. or one more thing is use the id and pass sent by client to be forwarded to my actual server and then its up to the server to authenticate it and connect the call or to reply with incorrect id and pass. this can be done only if FS could accept all registrations without it being defined in its profile and some how the dialplan was crafted to use the registration id and pass sent by client to be used as the id and pass for the SIP TRUNK or the server in general. -- View this message in context: http://www.nabble.com/how-to-do-upper-registration-tp22770442p22776631.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From steveu at coppice.org Mon Mar 30 04:48:03 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 30 Mar 2009 19:48:03 +0800 Subject: [Freeswitch-users] echo cancellation on PRI cards In-Reply-To: <5c7d82f20903291920od56993eu7a30a760e0ed4847@mail.gmail.com> References: <4cd9d780903162155l6538c42fx4ae531f7377e4c8f@mail.gmail.com> <49C03067.7070406@3c.co.uk> <49C03F5D.9050904@coppice.org> <49C05641.7070309@3c.co.uk> <49C0F05C.5090204@3c.co.uk> <49C0F9EA.3000200@freeswitch.org> <49CE29D1.3000603@3c.co.uk> <5c7d82f20903282230y57895d66hb607845415bd3c1d@mail.gmail.com> <49CF3238.9040503@coppice.org> <5c7d82f20903291920od56993eu7a30a760e0ed4847@mail.gmail.com> Message-ID: <49D0B173.9080002@coppice.org> Fadil Berisha wrote: > > Sure you can get reliable DTMF detection on 70%-80% of call paths with > no echo cancellation, > > > Fair enough. You say "sure you can get reliable DTMF detection on > 70%-80% of call paths with no echo cancellation". OK, you forgot to > mention that this is achievable with text-book algorithms and that > exist advanced algorithms with reliability close to 100%. From my > point of view , no need further arguing this issue and this thread for > me is closed. Could you enlighten us as to these advanced algorithms that beat conventional statistics? You keep referring to mystical adaptive algorithms on the OSLEC mailing list, but you never give any details there, either. Steve From anthony.minessale at gmail.com Mon Mar 30 05:53:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 07:53:39 -0500 Subject: [Freeswitch-users] upper registration in FS? In-Reply-To: <22764757.post@talk.nabble.com> References: <22764757.post@talk.nabble.com> Message-ID: <191c3a030903300553s5e2f371dk55a9c135600972a9@mail.gmail.com> FS does not proxy registrations, really it doesn't proxy anything very much besides calls in a limited capacity, you would want to use a SIP proxy for that. On Sun, Mar 29, 2009 at 1:07 AM, xbipin wrote: > > i have been trying to do this but seem to have been lost, actually i want > to > use freeswitch as a session border controller so basically all the clients > that try to register to FS will actually be authenticated by voipswitch but > i want FS to be in between and proxy everything and with any of its > advanced > options turned off so as to simply work as topology hiding, proxy, SBC > only. > -- > View this message in context: > http://www.nabble.com/upper-registration-in-FS--tp22764757p22764757.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/29a99765/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 30 06:04:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:04:31 -0500 Subject: [Freeswitch-users] FS - MjSip no voice In-Reply-To: <20090329220921.227090@gmx.net> References: <20090329220921.227090@gmx.net> Message-ID: <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> You should press f8 to get more detailed output from FS. you also should capture more of the call, starting at line 192 you seem to be sending yourself a notify, not sure how you did that. you are not by any chance trying to call a registered endpoint using the FS ip together with @ are you? say you fs box is 1.2.3.4 and the phone is registered as 1000 If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you would use sofia/internal/1000%1.2.3.4 The % tells it to resolve the domain as a locally hosted domain and translate it to the registered contact instead of using dns. otherwise, enable debugging with f8 and reproduce your issue and capture *all* the output. On Sun, Mar 29, 2009 at 5:09 PM, wrote: > Hello everyone, > > I am trying to get FS working with the MjSip Java Sip-stack, the SipToSis > source and the normal one. Everything works well within my own network and > when using x-lite, but when it comes to making calls from MjSip to an > outside FS server I don't hear any voice - seems to be a NAT problem or some > kind of other MjSip problem. Registration works fine though and SIP messages > get through ok, but non of the UDP RTP ones. Would be great if someone could > advice me on how to do the setup correctly. > > The whole FS trace can be found here: http://pastebin.freeswitch.org/8029 > > The settings for MjSip are: > > "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090", > "transport_protocols=udp tcp","from_url=", > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > "bin_rat=rat","bin_vic=vic" > > > Thank you very much. > Best wishes, > Phil > > -- > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/017be9a0/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 30 06:07:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:07:19 -0500 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <29b888f80903300056t51828330s39bbf662fe2cd8c0@mail.gmail.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> <29b888f80903300056t51828330s39bbf662fe2cd8c0@mail.gmail.com> Message-ID: <191c3a030903300607m55d7dcd8u78e8ea9fc7b491cb@mail.gmail.com> libdingaling.c:1545 xmpp_connect() io error 2 7 It means there was an I/O error trying to connect it's probably because iksemel does not have srv support, if you are trying to connect to gtalk you have to manually specify the server as talk.google.com, the default client.xml is self-explanatory for connecting to gtalk and it's on the wiki too.. On Mon, Mar 30, 2009 at 2:56 AM, Moiz Chinoy wrote: > Hi, > > If there is no build for the tls with freeswitch on windows, can you > please guide how to test gtalk integration on windows. > > I have successfully compiled the tls with iksemel, there were couple > of errors but I managed to compile it. > > Now, I am also getting the error libdingaling.c:1545 xmpp_connect() io > error 2 7 > > What is this error about? > > There is a similar post on the mailing list regarding the above IO > error but can't find the reply for it. > > On Sat, Mar 28, 2009 at 8:03 PM, Michael Jerris wrote: > > please see my previous response in this thread. > > > > MIke > > > > On Mar 28, 2009, at 5:09 AM, Moiz Chinoy wrote: > > > >> I am trying it on windows. > >> Where to get gnutls-devel for windows? > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Regards, > Moiz Chinoy. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/01eeb0ac/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 30 06:12:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:12:05 -0500 Subject: [Freeswitch-users] how to do upper registration In-Reply-To: <22776631.post@talk.nabble.com> References: <22776631.post@talk.nabble.com> Message-ID: <191c3a030903300612g19fb50f6n3b10ba7e8f650282@mail.gmail.com> is this deja-vu, this is the 2nd thread with the same subject, i seem to be seeing multiple of every email you send you may want to check your mail client. like ken said, and me too in the other thread with the same exact email in it, use a proxy http://www.kamailio.org/ On Mon, Mar 30, 2009 at 3:21 AM, xbipin wrote: > > so what i make of it is that FS cant do upper registration or lets say it > wont proxy everything and wont use the registration details it receives > from > clients registering to it and use it to register clients to a different > server for which its proxying, or mayb its not meant to do so but then i > ask > can FS be configured to accept all registrations coming to it without each > userid and password being defined in the profiles or can FS be configured > to > simply accept all incomming calls without the need for clients to register > to it in which case it will fullfill my needs partially. > > or one more thing is use the id and pass sent by client to be forwarded to > my actual server and then its up to the server to authenticate it and > connect the call or to reply with incorrect id and pass. this can be done > only if FS could accept all registrations without it being defined in its > profile and some how the dialplan was crafted to use the registration id > and > pass sent by client to be used as the id and pass for the SIP TRUNK or the > server in general. > -- > View this message in context: > http://www.nabble.com/how-to-do-upper-registration-tp22770442p22776631.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/4217d439/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 30 06:13:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:13:36 -0500 Subject: [Freeswitch-users] Freeswitch Hunt Group In-Reply-To: <564d7da90903300128q70196f17r4834ec7ffb933fcd@mail.gmail.com> References: <564d7da90903300128q70196f17r4834ec7ffb933fcd@mail.gmail.com> Message-ID: <191c3a030903300613y11efe2bcyb69e46740eba5e45@mail.gmail.com> the h323 support in FS is brand new. It's lacking video support at all. You would have to provide funding to round up some developers to implement what you need. 2009/3/30 Christian Bourke > Hi, > > I would like to implement freeswitch as an H323 gatekeeper. I would like to > know if its possible for incoming H323 calls to go into a hunt group, if no > 'operators' are logged into the hunt group then the call will bounce to an > auto attendant and the H323 caller will be played a short video message then > freeswitch will disconnect the call. > > Does this sound possible with Freeswitch? > > Thanks, > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/91ed57d5/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 30 06:14:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:14:35 -0500 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues In-Reply-To: <1238402050.4364.274.camel@localhost.localdomain> References: <1238155504.4364.222.camel@localhost.localdomain> <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> <1238402050.4364.274.camel@localhost.localdomain> Message-ID: <191c3a030903300614g25257ba4i4b46e5ab0caffe7b@mail.gmail.com> On Mon, Mar 30, 2009 at 3:34 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > Thanks for your quick&clear answers. > > > On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote: > > > If you are using on-hook agents, it will place as many outbound calls > > as there are people waiting. > > If you are using off-hook agents it will just connect the first free > > agent. > > By "people waiting" you mean "calls in the queue" and not "agents > waiting" right? > correct > > > > > - Is there an [easy] way (with some javascript or similar) to > > "emulate" Asterisk's distribution strategies to agents (by > > amount of time without calls, total number of answered calls, > > round robing, ...) in this cases? > > > > > > Easiest way would be to write a patch in C to mod_fifo it'self or > > propose a bounty for features and see if you can get the change > > approved by the developers. > > I'll give the patch a try once I have a bit more practice with > FreeSWITCH (never even launched it yet) and its API. Any hints or > "quick-start" guides for module writing? > > Fran?ois. > > ok, stop by irc if you need any pointers > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/8324ece5/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 30 06:15:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:15:25 -0500 Subject: [Freeswitch-users] H323 supported devices In-Reply-To: References: <564d7da90903300130i40b723a2s64d3cbbe5eddcf63@mail.gmail.com> Message-ID: <191c3a030903300615m74a3e301pbeac813f41ada8c3@mail.gmail.com> perhaps we should stick to one thread per topic within the same 1 week span. 2009/3/30 Ken Rice > I?m not sure if the video support is complete but both freeswitch and > opal support video... > Now I also see another email you send asking about freeswitch being a > gatekeeper... FreeSWITCH is just another h323 end point, it does not do all > gatekeeper functions like keep up with what IP what h323 user is on... > > However, as on your other message it might be possible to do what you > want... My suggestion would be to set it up and try... There needs to be > more h323 testing anyway > > Ken > > > ------------------------------ > *From: *Christian Bourke > *Reply-To: * > *Date: *Mon, 30 Mar 2009 18:30:49 +1000 > *To: * > *Subject: *Re: [Freeswitch-users] H323 supported devices > > Thank you Ken. Do you know if mod_opal supports both H323 voice and video? > > 2009/3/30 Ken Rice > > Yes its possible... Look at mod_opal > > > > ------------------------------ > *From: *Christian Bourke > *Reply-To: * > *Date: *Mon, 30 Mar 2009 10:55:14 +1000 > *To: * > *Subject: *[Freeswitch-users] H323 supported devices > > > Hi, > > I would like to implement freeswitch as a H323 gateway. I would like to be > able to receive calls from H323 devices such as Polycom V500 and Dlink > DVC-1000. > > Does anyone know if this is possible? > > > ------------------------------ > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/898ae964/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 30 06:24:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 08:24:49 -0500 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <49CF0C3C.8070508@xbipin.com> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> <49CF0C3C.8070508@xbipin.com> Message-ID: <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> I'm really starting to feel like we're playing musical threads here. On Sun, Mar 29, 2009 at 12:50 AM, Bipin Patel wrote: > hi, > > what are the other steps if done on windows platform with FS as SBC and > Voipswitch as the server. > secondly how to make the clients registering to voipswitch go through FS > in which case FS should simply accept all registrations but it should do > a forward registration to voipswitch and if voipswitch rejects then FS > should also reject. basically i need FS as a SBC with topology hiding. > > Regards, > Bipin > www.xbipin.com > +971-55-9270058 > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] freeswitch as a session border controller > From: Saeed Ahmed > To: freeswitch-users at lists.freeswitch.org > Date: Sunday, March 29, 2009 3:49:53 AM > > > http://wiki.freeswitch.org/wiki/SBC_Setup written by rod. > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > xbipin > > Sent: Saturday, March 28, 2009 8:22 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] freeswitch as a session border controller > > > > > > hi, > > > > im trying to do the same, use FS as a plain and simple SBC but cant > figure > > out how to do so > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/97dcccca/attachment-0002.html From steveu at coppice.org Mon Mar 30 06:56:18 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 30 Mar 2009 21:56:18 +0800 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> <49CF0C3C.8070508@xbipin.com> <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> Message-ID: <49D0CF82.7060009@coppice.org> Anthony Minessale wrote: > I'm really starting to feel like we're playing musical threads here. Just avoid playing them through low bit rate codecs. :-) Steve From dave at 3c.co.uk Mon Mar 30 07:14:21 2009 From: dave at 3c.co.uk (David Knell) Date: Mon, 30 Mar 2009 08:14:21 -0600 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <49D0CF82.7060009@coppice.org> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> <49CF0C3C.8070508@xbipin.com> <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> <49D0CF82.7060009@coppice.org> Message-ID: <49D0D3BD.5070602@3c.co.uk> Steve Underwood wrote: > Anthony Minessale wrote: > >> I'm really starting to feel like we're playing musical threads here. >> > Just avoid playing them through low bit rate codecs. :-) > I think we need an echo canceller ;-) --Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/985933e8/attachment-0002.html From freeswitch at servercorps.com Mon Mar 30 07:17:00 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Mon, 30 Mar 2009 09:17:00 -0500 Subject: [Freeswitch-users] freeswitch as a session border controller In-Reply-To: <49D0D3BD.5070602@3c.co.uk> References: <424bdbb90804171312v1bbe4cc2hf71da1cf3bf2346a@mail.gmail.com> <22760426.post@talk.nabble.com> <6898E3E671324610A4B05A87CA4B1387@SaeedLaptop> <49CF0C3C.8070508@xbipin.com> <191c3a030903300624p7efb5e8cua098f738aa1bf630@mail.gmail.com> <49D0CF82.7060009@coppice.org> <49D0D3BD.5070602@3c.co.uk> Message-ID: <92e7d2090903300717m2e620117m1653624ed19a8eea@mail.gmail.com> GAAAAHH /me gouges eyes out with EC card 2009/3/30 David Knell : > Steve Underwood wrote: > > Anthony Minessale wrote: > > > I'm really starting to feel like we're playing musical threads here. > > > Just avoid playing them through low bit rate codecs. :-) > > > I think we need an echo canceller ;-) > > --Dave > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From bipin at xbipin.com Mon Mar 30 07:33:26 2009 From: bipin at xbipin.com (xbipin) Date: Mon, 30 Mar 2009 07:33:26 -0700 (PDT) Subject: [Freeswitch-users] live iso image with freeswitch Message-ID: <22784622.post@talk.nabble.com> can any1 tell me where can i find a live cd image with the basic stuff to run FS and FS with all it tools installed and WITH A GUI, something like a pbx in a flash iso image so windows users like me find it easier to get testing with FS as the support for windows SIP proxy or any SIP related tool for windows platform is just about nil so i realized FS on windows also wont make much sense coz the rest of the developers etc use linux for FS and if i simply keep waiting for FS to actually do something productive on windows platform then it might take long or forever. if any i can provide me a live CD image with just enough tools to run FS to its fullest coz till date i have been only using Voipswitch and its time i need to implement TLS or any such type of encryption to reach new markets. -- View this message in context: http://www.nabble.com/live-iso-image-with-freeswitch-tp22784622p22784622.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Mar 30 07:43:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 09:43:27 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22784622.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> Message-ID: <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> On Mar 30, 2009, at 9:33 AM, xbipin wrote: > can any1 tell me where can i find a live cd image with the basic > stuff to run > FS and FS with all it tools installed and WITH A GUI, something like > a pbx > in a flash iso image so windows users like me find it easier to get > testing > with FS as the support for windows SIP proxy or any SIP related tool > for > windows platform is just about nil so i realized FS on windows also > wont > make much sense coz the rest of the developers etc use linux for FS > and if i > simply keep waiting for FS to actually do something productive on > windows > platform then it might take long or forever. Well this is a tall order... You know people ask for it.. or shall I say demand it... but nobody really steps up to help out at all on the GUI requests. FreeSWITCH on windows is equally capable minus a couple of things like TLS since nobody will actually DO the work required to make it happen. This isn't a buffet where you pull up and demand things be one way or the other... this is a community where you start helping. I would love to see more helping and less demanding! > if any i can provide me a live CD image with just enough tools to > run FS to > its fullest coz till date i have been only using Voipswitch and its > time i > need to implement TLS or any such type of encryption to reach new > markets. You could follow the linux how to and install CentOS and be done already. Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/11d6eac6/attachment-0002.html From gmaruzz at celliax.org Mon Mar 30 07:44:01 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 16:44:01 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22784622.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> Message-ID: <7b197bef0903300744p7b388be2x2ce1b30841722e3@mail.gmail.com> There is none yet available. If you have patience, I suspect that one will be out in the next weeks, tough. Watch the website and the mailing list for announcement. 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 Mon, Mar 30, 2009 at 4:33 PM, xbipin wrote: > > can any1 tell me where can i find a live cd image with the basic stuff to run > FS and FS with all it tools installed and WITH A GUI, something like a pbx > in a flash iso image so windows users like me find it easier to get testing > with FS as the support for windows SIP proxy or any SIP related tool for > windows platform is just about nil so i realized FS on windows also wont > make much sense coz the rest of the developers etc use linux for FS and if i > simply keep waiting for FS to actually do something productive on windows > platform then it might take long or forever. > > if any i can provide me a live CD image with just enough tools to run FS to > its fullest coz till date i have been only using Voipswitch and its time i > need to implement TLS or any such type of encryption to reach new markets. > -- > View this message in context: http://www.nabble.com/live-iso-image-with-freeswitch-tp22784622p22784622.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Mon Mar 30 07:44:58 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 30 Mar 2009 10:44:58 -0400 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22784622.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> Message-ID: On Mar 30, 2009, at 10:33 AM, xbipin wrote: > > can any1 tell me where can i find a live cd image with the basic > stuff to run > FS and FS with all it tools installed and WITH A GUI, something like > a pbx > in a flash iso image so windows users like me find it easier to get > testing > with FS as the support for windows SIP proxy or any SIP related tool > for > windows platform is just about nil so i realized FS on windows also > wont > make much sense coz the rest of the developers etc use linux for FS > and if i > simply keep waiting for FS to actually do something productive on > windows > platform then it might take long or forever. There is not currently a fully working gui or a live cd image. I think your comments about FreeSWITCH on windows are not correct. > if any i can provide me a live CD image with just enough tools to > run FS to > its fullest coz till date i have been only using Voipswitch and its > time i > need to implement TLS or any such type of encryption to reach new > markets. There is a live cd in the works, I am sure it will be announced here when it is ready. Mike From msc at freeswitch.org Mon Mar 30 08:23:59 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 30 Mar 2009 08:23:59 -0700 Subject: [Freeswitch-users] differences between mod_fifo and asterisk queues In-Reply-To: <1238402050.4364.274.camel@localhost.localdomain> References: <1238155504.4364.222.camel@localhost.localdomain> <191c3a030903270855j65b8babw5e33d84b4aa9d5d0@mail.gmail.com> <1238402050.4364.274.camel@localhost.localdomain> Message-ID: On Mar 30, 2009, at 1:34 AM, Fran?ois Delawarde wrote: > Thanks for your quick&clear answers. > > > On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote: > >> If you are using on-hook agents, it will place as many outbound calls >> as there are people waiting. >> If you are using off-hook agents it will just connect the first free >> agent. > > By "people waiting" you mean "calls in the queue" and not "agents > waiting" right? > >> >> - Is there an [easy] way (with some javascript or similar) to >> "emulate" Asterisk's distribution strategies to agents (by >> amount of time without calls, total number of answered calls, >> round robing, ...) in this cases? >> >> >> Easiest way would be to write a patch in C to mod_fifo it'self or >> propose a bounty for features and see if you can get the change >> approved by the developers. > > I'll give the patch a try once I have a bit more practice with > FreeSWITCH (never even launched it yet) and its API. Any hints or > "quick-start" guides for module writing? > There are some developer docs on the wiki. Nothing truly comprehensive but enough to get you going. Also see mod_skel in the source tree. -MC From bipin at xbipin.com Mon Mar 30 09:18:06 2009 From: bipin at xbipin.com (xbipin) Date: Mon, 30 Mar 2009 09:18:06 -0700 (PDT) Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> Message-ID: <22786890.post@talk.nabble.com> hi, im really sorry if it sounded as if i was demanding, its a place where every1 helps so i did the same thing, asked if any1 knew of any such live cd image coz its been quiet some time and i havent reached anywhere with the testing of FS coz the reason being, i can test it offline at my home but that will be like a single client connecting to FS and again the issue being, i live in a country where VoIP is blocked so cant even get FS to register to a provider also. then how do i ever test it, so in reply i would say i have a dedicated server and a proper VoIP setup with loads of clients but i have been using Voipswitch on windows till date and they got a very simple form of encryption which helps me bypass blocks and do the selling as well as testing, so now if i install FS on my server then how can i still test it in my existing setup coz the TLS, as discussed, is currently not functional on the windows platform but works on linux which is like learning a new thing from scratch, not that i cant, but will take up a lot of my time to reach to a level where a normal VoIP user on linux is currently compared to a windows user like me. in this part of the world, there r many guys selling Voip etc but all they know is of voipswitch as windows is what ppl know in this region. mayb i might be the only one from the whole region mostly using and promoting open source softwares like pfsense, opensbc and many such stuff and also might be the only one who knows there r things like asterisk and freeswitch. i searched like 75 outlets to get asterisk cards and all i got for an answer is, what is asterisk. living in this region and being able to test and try open source softwares is not easy coz never 2 guys meet and end up talking something about linux, pbx, softswitches etc coz no1 has ever heard of such things other than the stuff whats running in the economy. Brian West-3 wrote: > > > On Mar 30, 2009, at 9:33 AM, xbipin wrote: >> can any1 tell me where can i find a live cd image with the basic >> stuff to run >> FS and FS with all it tools installed and WITH A GUI, something like >> a pbx >> in a flash iso image so windows users like me find it easier to get >> testing >> with FS as the support for windows SIP proxy or any SIP related tool >> for >> windows platform is just about nil so i realized FS on windows also >> wont >> make much sense coz the rest of the developers etc use linux for FS >> and if i >> simply keep waiting for FS to actually do something productive on >> windows >> platform then it might take long or forever. > > Well this is a tall order... You know people ask for it.. or shall I > say demand it... but nobody really steps up to help out at all on the > GUI requests. > > FreeSWITCH on windows is equally capable minus a couple of things like > TLS since nobody will actually DO the work required to make it > happen. This isn't a buffet where you pull up and demand things be > one way or the other... this is a community where you start helping. > I would love to see more helping and less demanding! > >> if any i can provide me a live CD image with just enough tools to >> run FS to >> its fullest coz till date i have been only using Voipswitch and its >> time i >> need to implement TLS or any such type of encryption to reach new >> markets. > > > You could follow the linux how to and install CentOS and be done > already. > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/live-iso-image-with-freeswitch-tp22784622p22786890.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Mar 30 09:26:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Mar 2009 09:26:38 -0700 Subject: [Freeswitch-users] [Remote SIP client] Couple of questions In-Reply-To: <7.0.1.0.2.20090325104634.02701c88@fredshack.com> References: <7.0.1.0.2.20090325104634.02701c88@fredshack.com> Message-ID: <87f2f3b90903300926n7f387eb2rc17a04463fafa4c9@mail.gmail.com> Just following up... did you get these questions ironed out? -MC On Wed, Mar 25, 2009 at 2:51 AM, Gilles wrote: > Hello, > > I have a couple of questions related to having SIP users connecting > from the Net to a Freeswitch server through NAT routers on both ends: > > 1. How must I configure routers on both ends? I understand that I > need to route incoming TCP/UDP 5080 into the Freeswitch server, but > what about the other router? I guess I also need to route this port > to let the SIP phone ring, but what about data (RTP/RTCP)? > > 2. The Freeswitch server is connected to the POTS with either an > OpenVox PCI card or a Linksys 3102 box: When a call is made between > the POTS and a remote SIP phone (ie. out there on the Net, not on the > same LAN as the Freeswitch server), is there a way for data to flow > directly from the POTS to the remote SIP client instead of through > the Freeswitch server? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sicfslist at gmail.com Mon Mar 30 09:29:23 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 30 Mar 2009 11:29:23 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22786890.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> Message-ID: <35b355e90903300929l17b0ac5ax67696d77b4fac4a7@mail.gmail.com> Here is a really good start --> http://wiki.freeswitch.org/wiki/SBC_Setup Overall I think you're going to just have to install Linux on a box and get after it. It may be more painful in the short term ... but in the long term your life will be much better. The devs / community here are pretty incredible with their desire and efforts to help everyone (and all platforms) ... but in all reality it's a huge task. If you stick with Linux / FS everything will just work and there is a tremendous amount of resources on the web. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/9aa7c85a/attachment-0002.html From msc at freeswitch.org Mon Mar 30 09:33:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Mar 2009 09:33:02 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <22786890.post@talk.nabble.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> Message-ID: <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> > living in this region and being able to test and try open source softwares > is not easy coz never 2 guys meet and end up talking something about linux, > pbx, softswitches etc coz no1 has ever heard of such things other than the > stuff whats running in the economy. > Out of curiosity, what part of the world are you in? -MC From bipin at xbipin.com Mon Mar 30 10:26:25 2009 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 30 Mar 2009 21:26:25 +0400 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> Message-ID: <49D100C1.8010408@xbipin.com> hi, i currently live in a country called UAE - united arab emirates and a city called Dubai. Regards, Bipin -------- Original Message -------- Subject: Re: [Freeswitch-users] live iso image with freeswitch From: Michael Collins To: freeswitch-users at lists.freeswitch.org Date: Monday, March 30, 2009 8:33:02 PM >> living in this region and being able to test and try open source softwares >> is not easy coz never 2 guys meet and end up talking something about linux, >> pbx, softswitches etc coz no1 has ever heard of such things other than the >> stuff whats running in the economy. >> > > Out of curiosity, what part of the world are you in? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > __________ NOD32 3975 (20090330) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > > > From brian at freeswitch.org Mon Mar 30 10:30:27 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 12:30:27 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <49D100C1.8010408@xbipin.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> Message-ID: <2819EE9D-BFEA-4C0E-91BE-2A698E28CA9E@freeswitch.org> That is one interesting city.... I wouldn't mind paying a visit but its a bit rich for my blood! ;) On Mar 30, 2009, at 12:26 PM, Bipin Patel wrote: > hi, > > i currently live in a country called UAE - united arab emirates and a > city called Dubai. > > > Regards, > Bipin Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/674e1b3a/attachment-0002.html From msc at freeswitch.org Mon Mar 30 10:36:40 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Mar 2009 10:36:40 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <49D100C1.8010408@xbipin.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> Message-ID: <87f2f3b90903301036u1c0d1c80g413db346d7cc2eef@mail.gmail.com> On Mon, Mar 30, 2009 at 10:26 AM, Bipin Patel wrote: > hi, > > i currently live in a country called UAE - united arab emirates and a > city called Dubai. > Hehe, Dubai is quite a popular place - even a number of us ignorant Americans have heard of it! :) We would love to see FreeSWITCH become more popular in Dubai since it is such an important business hub in the Arab world. Please keep checking back for updates on the subject of live CDs or ISO install images. They'll be ready sooner or later, hopefully sooner. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/ee6befdb/attachment-0002.html From kristian.kielhofner at gmail.com Mon Mar 30 10:47:36 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 30 Mar 2009 13:47:36 -0400 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> Message-ID: <2d9149cd0903301047l65f654a2wd057427191520626@mail.gmail.com> The AstLinux ISO with FreeSWITCH is a live cd. http://mirror.astlinux.org/freeswitch/ This one is a little old but I could easily compile a new one... 2009/3/30 Brian West : > > On Mar 30, 2009, at 9:33 AM, xbipin wrote: > > can any1 tell me where can i find a live cd image with the basic stuff to > run > FS and FS with all it tools installed and WITH A GUI, something like a pbx > in a flash iso image so windows users like me find it easier to get testing > with FS as the support for windows SIP proxy or any SIP related tool for > windows platform is just about nil so i realized FS on windows also wont > make much sense coz the rest of the developers etc use linux for FS and if i > simply keep waiting for FS to actually do something productive on windows > platform then it might take long or forever. > > Well this is a tall order... You know people ask for it.. or shall I say > demand it... but nobody really steps up to help out at all on the GUI > requests. > FreeSWITCH on windows is equally capable minus a couple of things like TLS > since nobody will actually DO the work required to make it happen. ?This > isn't a buffet where you pull up and demand things be one way or the > other... this is a community where you start helping. ?I would love to see > more helping and less demanding! > > if any i can provide me a live CD image with just enough tools to run FS to > its fullest coz till date i have been only using Voipswitch and its time i > need to implement TLS or any such type of encryption to reach new markets. > > You could follow the linux how to and install CentOS and be done already. > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From gmaruzz at celliax.org Mon Mar 30 10:58:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 19:58:36 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <87f2f3b90903301036u1c0d1c80g413db346d7cc2eef@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <87f2f3b90903301036u1c0d1c80g413db346d7cc2eef@mail.gmail.com> Message-ID: <7b197bef0903301058m63f5c3afr74dd6b2c07f88a41@mail.gmail.com> Hi all, What I would like to stress is: 1) FreeSwitch is working on Windows, natively, without hacks 2) This is a huge advantage for a free software that want to be *really* popular (eg: be capable of running on an already working office machine, without dedicated hardware/expertise) 3) This is very important for people that are not "hard core", but just enthusiast, or just "wannabe". Why they have to go for a proprietary solution, maybe cracked? 4) This is very important for people/situation that just cannot afford another nmachine, or to dedicate a machine 5) Freeswitch is tested on Windows, albeit less than on *nix 6) This gap will be closing as the curve of adoption go further up 7) The efforts toward a GUI are proceeding, and in a multiplatform way, so they'll be working on Windows too I'm a *nix guy like you all, but let's bring closer to us, and us closer to, the vast majority of people/situations in the world. Especially making mind at the fact that the big effort, building FS multiplatform since the beginning, has been YET done :-) ! Ah, 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 2009/3/30 Michael Collins : > On Mon, Mar 30, 2009 at 10:26 AM, Bipin Patel wrote: >> >> hi, >> >> i currently live in a country called UAE - united arab emirates and a >> city called Dubai. > > Hehe, Dubai is quite a popular place - even a number of us ignorant > Americans have heard of it! :) We would love to see FreeSWITCH become more > popular in Dubai since it is such an important business hub in the Arab > world. Please keep checking back for updates on the subject of live CDs or > ISO install images. They'll be ready sooner or later, hopefully sooner. :) > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From gmaruzz at celliax.org Mon Mar 30 10:59:18 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 19:59:18 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <2d9149cd0903301047l65f654a2wd057427191520626@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <2d9149cd0903301047l65f654a2wd057427191520626@mail.gmail.com> Message-ID: <7b197bef0903301059i2683cf82xf078e9331e287d90@mail.gmail.com> Yes Kristian, please! On Mon, Mar 30, 2009 at 7:47 PM, Kristian Kielhofner wrote: > The AstLinux ISO with FreeSWITCH is a live cd. > > http://mirror.astlinux.org/freeswitch/ > > This one is a little old but I could easily compile a new one... > > 2009/3/30 Brian West : >> >> On Mar 30, 2009, at 9:33 AM, xbipin wrote: >> >> can any1 tell me where can i find a live cd image with the basic stuff to >> run >> FS and FS with all it tools installed and WITH A GUI, something like a pbx >> in a flash iso image so windows users like me find it easier to get testing >> with FS as the support for windows SIP proxy or any SIP related tool for >> windows platform is just about nil so i realized FS on windows also wont >> make much sense coz the rest of the developers etc use linux for FS and if i >> simply keep waiting for FS to actually do something productive on windows >> platform then it might take long or forever. >> >> Well this is a tall order... You know people ask for it.. or shall I say >> demand it... but nobody really steps up to help out at all on the GUI >> requests. >> FreeSWITCH on windows is equally capable minus a couple of things like TLS >> since nobody will actually DO the work required to make it happen. ?This >> isn't a buffet where you pull up and demand things be one way or the >> other... this is a community where you start helping. ?I would love to see >> more helping and less demanding! >> >> if any i can provide me a live CD image with just enough tools to run FS to >> its fullest coz till date i have been only using Voipswitch and its time i >> need to implement TLS or any such type of encryption to reach new markets. >> >> You could follow the linux how to and install CentOS and be done already. >> Brian West >> brian at freeswitch.org >> -- Meet us a ClueCon! ?http://www.cluecon.com >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mszlazak at aol.com Mon Mar 30 11:18:43 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Mon, 30 Mar 2009 14:18:43 -0400 Subject: [Freeswitch-users] Help scripting tone_detect. Message-ID: <8CB7F7B181E2057-1524-13D6@WEBMAIL-MZ40.sysops.aol.com> Except for fields in a dial plan extension, I can't get tone_detect (or stop_tone_detect), http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_stop_tone_detect to work in JavaScript. If there is one, what's the correct syntax for scripting these? Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/d3d6f2fa/attachment-0002.html From gkuri at ieee.org Mon Mar 30 11:31:16 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Mon, 30 Mar 2009 11:31:16 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <49D100C1.8010408@xbipin.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> Message-ID: <49D10FF4.7090501@ieee.org> eh, how are poeple doing VoIP over there, given use of it outside the UAE is "officially" outlawed by the TRA ? I've heard Etisalat is pretty strict with making sure it's blocked going outside the country via a L7 packet inspection device to drop SIP. ~Gabe Bipin Patel wrote: > hi, > > i currently live in a country called UAE - united arab emirates and a > city called Dubai. > > > Regards, > Bipin > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] live iso image with freeswitch > From: Michael Collins > To: freeswitch-users at lists.freeswitch.org > Date: Monday, March 30, 2009 8:33:02 PM > >>> living in this region and being able to test and try open source softwares >>> is not easy coz never 2 guys meet and end up talking something about linux, >>> pbx, softswitches etc coz no1 has ever heard of such things other than the >>> stuff whats running in the economy. >>> >> Out of curiosity, what part of the world are you in? >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> __________ NOD32 3975 (20090330) Information __________ >> >> This message was checked by NOD32 antivirus system. >> http://www.eset.com >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Mar 30 11:35:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 13:35:20 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <49D10FF4.7090501@ieee.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> Message-ID: <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> Really hard to inspect packets when they run on port 443 and are encrypted :P /b On Mar 30, 2009, at 1:31 PM, Gabriel Kuri wrote: > eh, how are poeple doing VoIP over there, given use of it outside the > UAE is "officially" outlawed by the TRA ? I've heard Etisalat is > pretty > strict with making sure it's blocked going outside the country via a > L7 > packet inspection device to drop SIP. > > ~Gabe Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/9978d78e/attachment-0002.html From telles-listas at devel-it.com.br Mon Mar 30 11:35:50 2009 From: telles-listas at devel-it.com.br (Rodrigo P. Telles) Date: Mon, 30 Mar 2009 15:35:50 -0300 Subject: [Freeswitch-users] Response/reply match Nated endpoints Message-ID: <49D11106.4040409@devel-it.com.br> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/0fcaf2d0/attachment-0002.html From kristian.kielhofner at gmail.com Mon Mar 30 11:41:30 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 30 Mar 2009 14:41:30 -0400 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> Message-ID: <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> Are you saying they should configure SIP TLS to run on port 443? :) 2009/3/30 Brian West : > Really hard to inspect packets when they run on port 443 and are encrypted > :P > /b -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From gkuri at ieee.org Mon Mar 30 11:48:16 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Mon, 30 Mar 2009 11:48:16 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> Message-ID: <49D113F0.7030204@ieee.org> yeah, obviously they'd have to enable TLS, but my question is more in terms of trying to use the mainstream providers outside the country, particularly for call termination/origination, which don't support TLS. ~Gabe Brian West wrote: > Really hard to inspect packets when they run on port 443 and are > encrypted :P > > /b > > On Mar 30, 2009, at 1:31 PM, Gabriel Kuri wrote: > >> eh, how are poeple doing VoIP over there, given use of it outside the >> UAE is "officially" outlawed by the TRA ? I've heard Etisalat is pretty >> strict with making sure it's blocked going outside the country via a L7 >> packet inspection device to drop SIP. >> >> ~Gabe > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Mon Mar 30 11:47:23 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 20:47:23 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> Message-ID: <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> no one would do that! On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner wrote: > Are you saying they should configure SIP TLS to run on port 443? :) > > 2009/3/30 Brian West : >> Really hard to inspect packets when they run on port 443 and are encrypted >> :P >> /b > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Mon Mar 30 11:48:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 13:48:08 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> Message-ID: <0E217C73-B2CB-40F4-9CE6-6BACC04D2E02@freeswitch.org> Even better run OpenVPN on port 53 :P /b On Mar 30, 2009, at 1:41 PM, Kristian Kielhofner wrote: > Are you saying they should configure SIP TLS to run on port 443? :) > > 2009/3/30 Brian West : >> Really hard to inspect packets when they run on port 443 and are >> encrypted >> :P >> /b Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/3090fba9/attachment-0002.html From brian at freeswitch.org Mon Mar 30 11:49:14 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 13:49:14 -0500 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> Message-ID: You sure could to get around some oppression! :P /b On Mar 30, 2009, at 1:47 PM, Giovanni Maruzzelli wrote: > no one would do that! > > > On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner > wrote: >> Are you saying they should configure SIP TLS to run on port 443? :) >> >> 2009/3/30 Brian West : >>> Really hard to inspect packets when they run on port 443 and are >>> encrypted >>> :P >>> /b Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com From gmaruzz at celliax.org Mon Mar 30 11:51:43 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 30 Mar 2009 20:51:43 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> Message-ID: <7b197bef0903301151lb97a157ua11bff974708213b@mail.gmail.com> ;-) On Mon, Mar 30, 2009 at 8:49 PM, Brian West wrote: > You sure could to get around some oppression! :P > > /b > > On Mar 30, 2009, at 1:47 PM, Giovanni Maruzzelli wrote: > >> no one would do that! >> >> >> On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner >> wrote: >>> Are you saying they should configure SIP TLS to run on port 443? :) >>> >>> 2009/3/30 Brian West : >>>> Really hard to inspect packets when they run on port 443 and are >>>> encrypted >>>> :P >>>> /b > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From can_man at gmx.de Mon Mar 30 13:33:53 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Mon, 30 Mar 2009 22:33:53 +0200 Subject: [Freeswitch-users] FS - MjSip no voice In-Reply-To: <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> References: <20090329220921.227090@gmx.net> <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> Message-ID: <20090330203353.59630@gmx.net> Hallo, thank you for your answer Anthony. > > starting at line 192 you seem to be sending yourself a notify, not sure > how you did that. That is indeed strange, I have looked at the MjSip code but haven't found the cause yet. > you are not by any chance trying to call a registered endpoint using the > FS > ip together with @ are you? > say you fs box is 1.2.3.4 and the phone is registered as 1000 > > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you > would > use sofia/internal/1000%1.2.3.4 > The % tells it to resolve the domain as a locally hosted domain and > translate it to the registered contact instead of using dns. > For testing I at the moment send the incoming call to the voicemail of user 1000 with this code: return '''\n'''\ '''\n'''\ '''
\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''\n'''\ '''
\n'''\ '''''' % (didNumber, didNumber, id) Works fine with a normal SIP client. I have captured more output with debug enabled and have also captured the SIP messages originating from MjSip. FS: http://pastebin.freeswitch.org/8045 MjSip: http://pastebin.freeswitch.org/8046 Thank you very much for your help. Best wishes, Phil > > > On Sun, Mar 29, 2009 at 5:09 PM, wrote: > > > Hello everyone, > > > > I am trying to get FS working with the MjSip Java Sip-stack, the > SipToSis > > source and the normal one. Everything works well within my own network > and > > when using x-lite, but when it comes to making calls from MjSip to an > > outside FS server I don't hear any voice - seems to be a NAT problem or > some > > kind of other MjSip problem. Registration works fine though and SIP > messages > > get through ok, but non of the UDP RTP ones. Would be great if someone > could > > advice me on how to do the setup correctly. > > > > The whole FS trace can be found here: > http://pastebin.freeswitch.org/8029 > > > > The settings for MjSip are: > > > > "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090", > > "transport_protocols=udp tcp","from_url=", > > > > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > > > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > > > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > > "bin_rat=rat","bin_vic=vic" > > > > > > Thank you very much. > > Best wishes, > > Phil > > > > -- > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 -- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss f?r nur 17,95 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a From anthony.minessale at gmail.com Mon Mar 30 14:20:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 16:20:19 -0500 Subject: [Freeswitch-users] FS - MjSip no voice In-Reply-To: <20090330203353.59630@gmx.net> References: <20090329220921.227090@gmx.net> <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> <20090330203353.59630@gmx.net> Message-ID: <191c3a030903301420l78f848ft281430e4d67daa0e@mail.gmail.com> maybe that phone does not support early media try adding the answer application to your dialplan On Mon, Mar 30, 2009 at 3:33 PM, wrote: > Hallo, > > thank you for your answer Anthony. > > > > > starting at line 192 you seem to be sending yourself a notify, not sure > > how you did that. > > That is indeed strange, I have looked at the MjSip code but haven't found > the cause yet. > > > you are not by any chance trying to call a registered endpoint using the > > FS > > ip together with @ are you? > > say you fs box is 1.2.3.4 and the phone is registered as 1000 > > > > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you > > would > > use sofia/internal/1000%1.2.3.4 > > The % tells it to resolve the domain as a locally hosted domain and > > translate it to the registered contact instead of using dns. > > > > For testing I at the moment send the incoming call to the voicemail of user > 1000 with this code: > > return '''\n'''\ > '''\n'''\ > '''
\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''\n'''\ > '''
\n'''\ > '''''' % (didNumber, didNumber, id) > > > Works fine with a normal SIP client. > I have captured more output with debug enabled and have also captured the > SIP messages originating from MjSip. > > FS: http://pastebin.freeswitch.org/8045 > MjSip: http://pastebin.freeswitch.org/8046 > > Thank you very much for your help. > Best wishes, > Phil > > > > > > > On Sun, Mar 29, 2009 at 5:09 PM, wrote: > > > > > Hello everyone, > > > > > > I am trying to get FS working with the MjSip Java Sip-stack, the > > SipToSis > > > source and the normal one. Everything works well within my own network > > and > > > when using x-lite, but when it comes to making calls from MjSip to an > > > outside FS server I don't hear any voice - seems to be a NAT problem or > > some > > > kind of other MjSip problem. Registration works fine though and SIP > > messages > > > get through ok, but non of the UDP RTP ones. Would be great if someone > > could > > > advice me on how to do the setup correctly. > > > > > > The whole FS trace can be found here: > > http://pastebin.freeswitch.org/8029 > > > > > > The settings for MjSip are: > > > > > > "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090", > > > "transport_protocols=udp tcp","from_url= >", > > > > > > > > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > > > > > > > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > > > > > > > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > > > "bin_rat=rat","bin_vic=vic" > > > > > > > > > Thank you very much. > > > Best wishes, > > > Phil > > > > > > -- > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > -- > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/db3ed669/attachment-0002.html From keithl at voxtelecom.co.za Mon Mar 30 14:56:57 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Mon, 30 Mar 2009 23:56:57 +0200 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. Message-ID: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> Hi, I have an application where my Javascript hanguphook code calculates a value (e.g. the cost of the call which can only be calculated post hangup) and I need to have that value appear as a field in the cdrs. As the 'session' object is no longer available for javascript logic post hangup, I can't figure out how to 'set' a user variable post hangup, such that it can be written to the cdr when the state changes from CS_HANGUP -> CS_REPORTING. Maybe it's just not possible......? It would be a pity to have to resort to writing out cdrs from the javascript itself and duplicating what fs does so well already. Any advice / suggestions would be appreciated. Best Regards Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/f80d0c7f/attachment-0002.html From msc at freeswitch.org Mon Mar 30 15:08:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Mar 2009 15:08:41 -0700 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> Message-ID: <87f2f3b90903301508v5f6a5200qd918aa1fa5f280a1@mail.gmail.com> Are logging both a- and b-legs? Just curious what your setup is. -MC 2009/3/30 Keith Laaks > Hi, > > > > I have an application where my Javascript hanguphook code calculates a > value (e.g. the cost of the call which can only be calculated post hangup) > and I need to have that value appear as a field in the cdrs. > > > > As the ?session? object is no longer available for javascript logic post > hangup, I can?t figure out how to ?set? a user variable post hangup, such > that it can be written to the cdr when the state changes from CS_HANGUP -> > CS_REPORTING. Maybe it?s just not possible??? It would be a pity to have to > resort to writing out cdrs from the javascript itself and duplicating what > fs does so well already. > > > > Any advice / suggestions would be appreciated. > > > > Best Regards > > > > Keith > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/520305c7/attachment-0002.html From anthony.minessale at gmail.com Mon Mar 30 15:12:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Mar 2009 17:12:58 -0500 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> Message-ID: <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> in your script called via api_hangup_hook: var env = request.dumpENV("text"); consoleLog("info", env); all those vars are there for you, you can get the individually with var hval = request.getHeader("some_header"); 2009/3/30 Keith Laaks > Hi, > > > > I have an application where my Javascript hanguphook code calculates a > value (e.g. the cost of the call which can only be calculated post hangup) > and I need to have that value appear as a field in the cdrs. > > > > As the ?session? object is no longer available for javascript logic post > hangup, I can?t figure out how to ?set? a user variable post hangup, such > that it can be written to the cdr when the state changes from CS_HANGUP -> > CS_REPORTING. Maybe it?s just not possible??? It would be a pity to have to > resort to writing out cdrs from the javascript itself and duplicating what > fs does so well already. > > > > Any advice / suggestions would be appreciated. > > > > Best Regards > > > > Keith > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/282a43bc/attachment-0002.html From dule.maillist at gmail.com Mon Mar 30 16:44:05 2009 From: dule.maillist at gmail.com (Dan Le) Date: Mon, 30 Mar 2009 19:44:05 -0400 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <49CC8DFE.3050104@gmx.net> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> Message-ID: <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> I get similar behavior as Peter when trying to enter a locked conference. If I am just dialing from a phone to a conference (on a dialplan), it will properly lock me out. But if I do an originate command (originate sofia/internal/1001 &conference(3000)), it will drop me into the conference, even though it is suppose to be locked. I am using the released 1.0.3 tag. On Fri, Mar 27, 2009 at 4:27 AM, Peter P GMX wrote: > It's defined via XML-Curl, and manual dialling and transfering do > trigger the same xml-curl request. This means that this conference > number is not defined in the any xml conf file. > If I transfer a call (without PIN) and then manually dial with another > phone into this conf with PIN, both calls are in the same conference. > > I have SVN rev 12796. > > > Best regards > Peter > > > Michael Collins schrieb: > > On Thu, Mar 26, 2009 at 4:09 PM, Peter P GMX > wrote: > > > >> Hello Michael, > >> > >> I tried this, but received the same behaviour. It does not ask for the > >> defined PIN. > >> > > > > Just curious - where do you define the PIN for this conference? > > -MC > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/fd360936/attachment-0002.html From brian at freeswitch.org Mon Mar 30 16:54:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 30 Mar 2009 18:54:08 -0500 Subject: [Freeswitch-users] Originate and Conference In-Reply-To: <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> References: <49CBEA8D.4050901@gmx.net> <87f2f3b90903261432n35c08a83v8d6c43246a2c28fd@mail.gmail.com> <49CC0B47.6000508@gmx.net> <87f2f3b90903261658ie748201lf8bd209f4827be79@mail.gmail.com> <49CC8DFE.3050104@gmx.net> <914fc92a0903301644y4a0f56e2ob039d15fb0654432@mail.gmail.com> Message-ID: <0C56F0E6-CB68-4C71-BB33-C31097D135CF@freeswitch.org> Update again to svn trunk... btw 1.0.4 pre3 is out on files.freeswitch.org /b On Mar 30, 2009, at 6:44 PM, Dan Le wrote: > I get similar behavior as Peter when trying to enter a locked > conference. > > If I am just dialing from a phone to a conference (on a dialplan), > it will properly lock me out. But if I do an originate command > (originate sofia/internal/1001 &conference(3000)), it will drop me > into the conference, even though it is suppose to be locked. > > I am using the released 1.0.3 tag. > Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090330/be7c929e/attachment-0002.html From bipin at xbipin.com Mon Mar 30 21:55:28 2009 From: bipin at xbipin.com (xbipin) Date: Mon, 30 Mar 2009 21:55:28 -0700 (PDT) Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <22786890.post@talk.nabble.com> <87f2f3b90903300933s71e40b06j5704ca4712446ee0@mail.gmail.com> <49D100C1.8010408@xbipin.com> <49D10FF4.7090501@ieee.org> <60403C01-A436-450E-9F28-827850D29F2B@freeswitch.org> <2d9149cd0903301141g206bf298n811ce559ad96a47d@mail.gmail.com> <7b197bef0903301147g6003d33bkfb3f5c08519c80ea@mail.gmail.com> Message-ID: <22798341.post@talk.nabble.com> hi, i think this thread has got a bit too much of attention but in all directions, bytheway sorry for that double posting but i think its something to do with nabble as i get a mail saying ur post is rejected so i resend and then i find both the posts get accepted. UAE is very strict in terms of allowing VoIP, they block skype and any sort of voip available but come to see it, they use all the latest gadgets to block advanced stuff but its basic stuff that bypasses it thats y guys like me end up selling voip and any1 dealing in UAE will only tell u one thing, its a million dollar market for voip. simple technic to bypass is, use any port other than the standard SIP ports and in the SIP packet, if the header says SIP or something like that, change it to SI_P or S_IP or whatever u like as the header and ur through the block. known methods to bypass r openvpn but they find out and simply block the port, XOR or CBCOM as said by grandstream, voipswitch's voiptunnel and just about anything that doesnt say anything about SIP or RTP in the packet header or description or whatever it may be. bytheway that astlinux is for embedded systems etc right? i have a ALIX board already running pfsense, im looking for a normal linux image for a workstation but just enough tools to run freeswitch, have the gnome or K desktop environment and later on when the web GUI is developed then simply the web GUI. -- View this message in context: http://www.nabble.com/live-iso-image-with-freeswitch-tp22784622p22798341.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From moizchinoy at gmail.com Mon Mar 30 23:00:05 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Tue, 31 Mar 2009 10:00:05 +0400 Subject: [Freeswitch-users] Error Compiling iksemel... In-Reply-To: <191c3a030903300607m55d7dcd8u78e8ea9fc7b491cb@mail.gmail.com> References: <29b888f80903252359m3f9be0aek4cb1bb6b6fa5b51c@mail.gmail.com> <1A5BED051CF6432A9FE2B7E842BB7ABD@quos> <29b888f80903260446t2def26fbyc9591fc4476e5ba1@mail.gmail.com> <1B117EFC18604C6E82663FDAEB3342A8@quos> <29b888f80903280209x69279523p8902a5b5ef2a338a@mail.gmail.com> <16E8D429-E5A1-479D-9AD3-14D332839F57@jerris.com> <29b888f80903300056t51828330s39bbf662fe2cd8c0@mail.gmail.com> <191c3a030903300607m55d7dcd8u78e8ea9fc7b491cb@mail.gmail.com> Message-ID: <29b888f80903302300q1eb097cey2178fded546984af@mail.gmail.com> This I/O error occurred once. And I am able to communicate google talk to SIP softphone through FS. Still does not know why this error happened as I did not change anything! 2009/3/30 Anthony Minessale : > libdingaling.c:1545 xmpp_connect() io error 2 7 > > It means there was an I/O error trying to connect > > it's probably because iksemel does not have srv support, if you are trying > to connect to gtalk > you have to manually specify the server as talk.google.com, the default > client.xml is self-explanatory for > connecting to gtalk and it's on the wiki too.. > > > On Mon, Mar 30, 2009 at 2:56 AM, Moiz Chinoy wrote: >> >> Hi, >> >> If there is no build for the tls with freeswitch on windows, can you >> please guide how to test gtalk integration on windows. >> >> I have successfully compiled the tls with iksemel, there were couple >> of errors but I managed to compile it. >> >> Now, I am also getting the error libdingaling.c:1545 xmpp_connect() io >> error 2 7 >> >> What is this error about? >> >> There is a similar post on the mailing list regarding the above IO >> error but can't find the reply for it. >> >> On Sat, Mar 28, 2009 at 8:03 PM, Michael Jerris wrote: >> > please see my previous response in this thread. >> > >> > MIke >> > >> > On Mar 28, 2009, at 5:09 AM, Moiz Chinoy wrote: >> > >> >> I am trying it on windows. >> >> Where to get gnutls-devel for windows? >> >> >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Regards, >> Moiz Chinoy. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Moiz Chinoy. From gmaruzz at celliax.org Tue Mar 31 00:26:57 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 31 Mar 2009 09:26:57 +0200 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To Message-ID: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS Microsoft Windows Download 118MB HD: http://www.celliax.org/final.avi 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 From jason at jasonjgw.net Tue Mar 31 00:28:58 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 31 Mar 2009 18:28:58 +1100 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> Message-ID: <20090331072858.GA15463@jdc.jasonjgw.net> Brian West wrote: > This isn't a buffet where you pull up and demand things be one way or the > other... this is a community where you start helping. I would love to > see more helping and less demanding! So would I. I regularly scan the mailing list looking for questions to answer, but many of them relate to scenarios of which I have had no experience, or features that I haven't had any reason to use. I am worried that the questions are answered by the same people much of the time, which in the long run will be bad for the project as the community grows, i.e., more technical support work for the same people (who are mostly the core developers as well) is not a sustainable proposition. How do other large projects handle this? Does anyone have any positive suggestions to offer that would encourage more contributions? From bipin at xbipin.com Tue Mar 31 01:22:16 2009 From: bipin at xbipin.com (xbipin) Date: Tue, 31 Mar 2009 01:22:16 -0700 (PDT) Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> Message-ID: <22800505.post@talk.nabble.com> hi, is it just audio or is it that im having broken codecs so cant view any video? Regards, Bipin Giovanni Maruzzelli-3 wrote: > > Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS > Microsoft Windows > Download 118MB HD: http://www.celliax.org/final.avi > > 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p22800505.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From regs at kinetix.gr Tue Mar 31 02:03:10 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 31 Mar 2009 12:03:10 +0300 Subject: [Freeswitch-users] killall -HUP freeswitch Message-ID: <49D1DC4E.7070002@kinetix.gr> Hi, I am doing a "killall -HUP freeswitch" in order to achieve cdr-csv log rotation and the process gets killed instead. A few days ago it worked fine. I also noticed that a few days ago when I hit Ctrl+C freeswitch did not exit. Now, when I do this the process gets terminated immediately. What could be the cause of all this? I am using the svn 12548 trunk. I did not do any recompiling of FS. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From regs at kinetix.gr Tue Mar 31 02:10:23 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 31 Mar 2009 12:10:23 +0300 Subject: [Freeswitch-users] killall -HUP freeswitch In-Reply-To: <49D1DC4E.7070002@kinetix.gr> References: <49D1DC4E.7070002@kinetix.gr> Message-ID: <49D1DDFF.2080402@kinetix.gr> I found out the cause : When the mod_opal module is loaded the FS process gets killed with a kill -HUP. I thought it would be good for everyone to know. Apostolos Pantsiopoulos wrote: > Hi, > > I am doing a "killall -HUP freeswitch" in order to achieve > cdr-csv log rotation and the process gets killed instead. A few > days ago it worked fine. I also noticed that a few days ago when I > hit Ctrl+C freeswitch did not exit. Now, when I do this the process gets > terminated immediately. What could be the cause of all this? I am using > the svn 12548 trunk. I did not do any recompiling of FS. > > > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From solko at gcdf.pl Tue Mar 31 03:05:57 2009 From: solko at gcdf.pl (Szymon Olko) Date: Tue, 31 Mar 2009 12:05:57 +0200 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <22800505.post@talk.nabble.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> Message-ID: <49D1EB05.7090804@gcdf.pl> xbipin pisze: > hi, > > is it just audio or is it that im having broken codecs so cant view any > video? > There are both, video and audio. Mplayer dump Otwieram dekoder video: [ffmpeg] FFmpeg's libavcodec codec family Wybrany kodek video: [ffcamtasia] vfm: ffmpeg (TechSmith Camtasia Screen Codec (native)) ========================================================================== ========================================================================== Otwieram dekoder audio: [pcm] Uncompressed PCM audio decoder AUDIO: 22050 Hz, 1 ch, s16le, 352.8 kbit/100.00% (ratio: 44100->44100) Wybrany kodek audio: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== > Regards, > Bipin > > > > Giovanni Maruzzelli-3 wrote: >> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS >> Microsoft Windows >> Download 118MB HD: http://www.celliax.org/final.avi >> >> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From keithl at voxtelecom.co.za Tue Mar 31 04:25:12 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Tue, 31 Mar 2009 13:25:12 +0200 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> Message-ID: <1B99233662E2104983E3550185D3ED73498352@xena.internal.datapro.co.za> Hi, Here is what I am trying to accomplish: //--- completecall.js trigged via api_hangup_hook ---- use("CURL"); const loglevel='notice'; var uuid = request.getHeader("Core-UUID"); var billmsec = request.getHeader("billmsec"); var urlrequest = "UUID=" + uuid + "&billmsec=" + billmsec; function reply_callback(string, arg) { string = string.substring(string.search("API")+3); string = string.substring(0,string.search("<")); var splits = string.split("~"); var i = 0; var length = splits.length; for (i=0; i < length; i++) { var fv=splits[i].split("="); console_log(loglevel, "setting: " + fv[0] + " = " + fv[1] + "\n"); //session.setVariable(fv[0],fv[1]); // Cant use set here as the session is dead by now - the call has been terminated } return true; } var curl = new CURL(); console_log(loglevel,"-- completecall.js ->" + url + "?" + urlrequest + "\n"); var env = request.dumpENV("text"); // debug console_log("ENV:\n", env + "\n"); // debug curl.run("POST", "http://127.0.0.1:10502/rate", urlrequest, reply_callback, "CBrate\n"); //returns string like APIcharge=10.20 exit(); So the trick is that I am accessing an external system via CURL where the call rate is calculated and returned (based on the call duration and uuid). Now I need to use a 'setVariable' to get it back as one of the parameters that the cdr module can write out for me. Note that in the above, the external system will return 'charge=value'. I need to set the variable 'charge' to the value in 'value'. Then using the config below, 'charge' can be written out as one of the cdr fields. Best Regards Keith From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: 31 March 2009 00:13 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Javascript, Hanguphooks,CDRs and User Variables. in your script called via api_hangup_hook: var env = request.dumpENV("text"); consoleLog("info", env); all those vars are there for you, you can get the individually with var hval = request.getHeader("some_header"); 2009/3/30 Keith Laaks Hi, I have an application where my Javascript hanguphook code calculates a value (e.g. the cost of the call which can only be calculated post hangup) and I need to have that value appear as a field in the cdrs. As the 'session' object is no longer available for javascript logic post hangup, I can't figure out how to 'set' a user variable post hangup, such that it can be written to the cdr when the state changes from CS_HANGUP -> CS_REPORTING. Maybe it's just not possible......? It would be a pity to have to resort to writing out cdrs from the javascript itself and duplicating what fs does so well already. Any advice / suggestions would be appreciated. Best Regards Keith _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/ff03685f/attachment-0002.html From anthony.minessale at gmail.com Tue Mar 31 05:59:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 07:59:34 -0500 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <1B99233662E2104983E3550185D3ED73498352@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> <1B99233662E2104983E3550185D3ED73498352@xena.internal.datapro.co.za> Message-ID: <191c3a030903310559g49211e66x3cbdc1dac088e671@mail.gmail.com> if you set the channel variable 'session_in_hangup_hook=true' early in the call, the session will be present in your script. 2009/3/31 Keith Laaks > Hi, > > > > Here is what I am trying to accomplish: > > > > //--- completecall.js trigged via api_hangup_hook ---- > > > > use("CURL"); > > > > const loglevel='notice'; > > var uuid = request.getHeader("Core-UUID"); > > var billmsec = request.getHeader("billmsec"); > > var urlrequest = "UUID=" + uuid + "&billmsec=" + billmsec; > > > > function reply_callback(string, arg) { > > string = string.substring(string.search("API")+3); > > string = string.substring(0,string.search("<")); > > var splits = string.split("~"); > > var i = 0; > > var length = splits.length; > > for (i=0; i < length; i++) { > > var fv=splits[i].split("="); > > console_log(loglevel, "setting: " + fv[0] + " = " + fv[1] + "\n"); > > //session.setVariable(fv[0],fv[1]); // Cant use set here as the session > is dead by now - the call has been terminated > > } > > return true; > > } > > > > var curl = new CURL(); > > console_log(loglevel,"-- completecall.js ->" + url + "?" + urlrequest + > "\n"); > > > > var env = request.dumpENV("text"); // debug > > console_log("ENV:\n", env + "\n"); // debug > > > > curl.run("POST", "http://127.0.0.1:10502/rate", urlrequest, > reply_callback, "CBrate\n"); > > //returns string like APIcharge=10.20 > > > > exit(); > > > > > > So the trick is that I am accessing an external system via CURL where the > call rate is calculated and returned (based on the call duration and uuid). > > Now I need to use a ?setVariable? to get it back as one of the parameters > that the cdr module can write out for me. > > Note that in the above, the external system will return ?charge=value?. > > I need to set the variable ?charge? to the value in ?value?. > > Then using the config below, ?charge? can be written out as one of the cdr > fields. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Best Regards > > > > Keith > > > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* 31 March 2009 00:13 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Javascript, Hanguphooks,CDRs and User > Variables. > > > > in your script called via api_hangup_hook: > > var env = request.dumpENV("text"); > > consoleLog("info", env); > > all those vars are there for you, you can get the individually with > var hval = request.getHeader("some_header"); > > 2009/3/30 Keith Laaks > > Hi, > > > > I have an application where my Javascript hanguphook code calculates a > value (e.g. the cost of the call which can only be calculated post hangup) > and I need to have that value appear as a field in the cdrs. > > > > As the ?session? object is no longer available for javascript logic post > hangup, I can?t figure out how to ?set? a user variable post hangup, such > that it can be written to the cdr when the state changes from CS_HANGUP -> > CS_REPORTING. Maybe it?s just not possible??? It would be a pity to have to > resort to writing out cdrs from the javascript itself and duplicating what > fs does so well already. > > > > Any advice / suggestions would be appreciated. > > > > Best Regards > > > > Keith > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/76a269c0/attachment-0002.html From andy at fabulous4.co.uk Tue Mar 31 06:04:19 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 31 Mar 2009 14:04:19 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message Message-ID: <30BF7571B7FC4F029C643D0E479CE614@wsandy> Hi, I'm using freeswitch as a glorified answering machine. FS registers with a VOIP gateway and all calls into the gateway go through an ivr menu and are allowed to leave a message which gets recorded to a file. The FS box is behind a NAT firewall. Everything works fine except that intermittently, calls keep getting cut off after a number of seconds. I've attached a snapshot of the log at the point that the call gets cut off, can anyone suggest why this is happening or how I can prevent it? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/81df0c39/attachment-0002.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: log-snippet.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/81df0c39/attachment-0002.txt From brian at freeswitch.org Tue Mar 31 06:26:41 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 08:26:41 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <30BF7571B7FC4F029C643D0E479CE614@wsandy> References: <30BF7571B7FC4F029C643D0E479CE614@wsandy> Message-ID: I'm going to guess you're not on SVN trunk? what rev are you on? /b On Mar 31, 2009, at 8:04 AM, Andy Ayers wrote: > Hi, > > I'm using freeswitch as a glorified answering machine. FS registers > with a VOIP gateway and all calls into the gateway go through an ivr > menu and are allowed to leave a message which gets recorded to a > file. The FS box is behind a NAT firewall. Everything works fine > except that intermittently, calls keep getting cut off after a > number of seconds. > > I've attached a snapshot of the log at the point that the call gets > cut off, can anyone suggest why this is happening or how I can > prevent it? > > Many thanks > Andy > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/0686c6e9/attachment-0002.html From andy at fabulous4.co.uk Tue Mar 31 06:36:29 2009 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 31 Mar 2009 14:36:29 +0100 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: Message-ID: <8B544F779B6948C9BF77DBE6D5599CFA@wsandy> Hi Brian, 1.03 Thanks Andy -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:27 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message I'm going to guess you're not on SVN trunk? what rev are you on? /b On Mar 31, 2009, at 8:04 AM, Andy Ayers wrote: Hi, I'm using freeswitch as a glorified answering machine. FS registers with a VOIP gateway and all calls into the gateway go through an ivr menu and are allowed to leave a message which gets recorded to a file. The FS box is behind a NAT firewall. Everything works fine except that intermittently, calls keep getting cut off after a number of seconds. I've attached a snapshot of the log at the point that the call gets cut off, can anyone suggest why this is happening or how I can prevent it? Many thanks Andy _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/264b1809/attachment-0002.html From brian at freeswitch.org Tue Mar 31 06:39:34 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 08:39:34 -0500 Subject: [Freeswitch-users] Calls being cut off while recording a message In-Reply-To: <8B544F779B6948C9BF77DBE6D5599CFA@wsandy> References: <8B544F779B6948C9BF77DBE6D5599CFA@wsandy> Message-ID: Please try SVN trunk. /b On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote: > Hi Brian, > > 1.03 > > Thanks > Andy Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/118180a9/attachment-0002.html From can_man at gmx.de Tue Mar 31 07:06:48 2009 From: can_man at gmx.de (can_man at gmx.de) Date: Tue, 31 Mar 2009 16:06:48 +0200 Subject: [Freeswitch-users] FS - MjSip no voice [SOLVED] SIP 200 / 183 problem In-Reply-To: <191c3a030903301420l78f848ft281430e4d67daa0e@mail.gmail.com> References: <20090329220921.227090@gmx.net> <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> <20090330203353.59630@gmx.net> <191c3a030903301420l78f848ft281430e4d67daa0e@mail.gmail.com> Message-ID: <20090331140648.227850@gmx.net> Hello, I have found the problem. FS on my local network sends "SIP/2.0 200 OK" after an invite and FS on the net through the external profil sends SIP/2.0 183 Session Progress. But MjSip doesn't know how to deal with 183, so it just ignores the message. For testing I have changed the 183 header to the 200 one and now it works. Thank you for your help and the quick response time. Best wishes, Phil >From FS on the net through the external profil: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 90.181.59.141:5090;rport=60315;branch=z9hG4bK256321;received=78.105.17.88 From: ;tag=z9hG4bK40977269 To: ;tag=vgg3Zja8pNQcg Call-ID: 507347917247 at 90.181.59.141 CSeq: 1 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12839M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 267 v=0 o=FreeSWITCH 1072777625698755085 8893522831081357051 IN IP4 90.181.59.141 s=FreeSWITCH c=IN IP4 91.121.59.148 t=0 0 m=audio 26722 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 >From FS in my local network: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.143:5060;rport=5060;branch=z9hG4bK423233;received=192.168.1.102 From: ;tag=z9hG4bK42598163 To: ;tag=Q0X494ZUNaKHH Call-ID: 961142687222 at 192.168.1.143 CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12712M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 267 v=0 o=FreeSWITCH 5195745633884389954 8941954824002056485 IN IP4 192.168.1.143 s=FreeSWITCH c=IN IP4 192.168.1.143 t=0 0 m=audio 22680 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 > maybe that phone does not support early media > > try adding the answer application to your dialplan > > > On Mon, Mar 30, 2009 at 3:33 PM, wrote: > > > Hallo, > > > > thank you for your answer Anthony. > > > > > > > > starting at line 192 you seem to be sending yourself a notify, not > sure > > > how you did that. > > > > That is indeed strange, I have looked at the MjSip code but haven't > found > > the cause yet. > > > > > you are not by any chance trying to call a registered endpoint using > the > > > FS > > > ip together with @ are you? > > > say you fs box is 1.2.3.4 and the phone is registered as 1000 > > > > > > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you > > > would > > > use sofia/internal/1000%1.2.3.4 > > > The % tells it to resolve the domain as a locally hosted domain and > > > translate it to the registered contact instead of using dns. > > > > > > > For testing I at the moment send the incoming call to the voicemail of > user > > 1000 with this code: > > > > return '''\n'''\ > > '''\n'''\ > > '''
\n'''\ > > '''\n'''\ > > '''\n'''\ > > ''' expression="^(%s)$">\n'''\ > > '''\n'''\ > > '''\n'''\ > > '''\n'''\ > > '''\n'''\ > > '''
\n'''\ > > '''''' % (didNumber, didNumber, id) > > > > > > Works fine with a normal SIP client. > > I have captured more output with debug enabled and have also captured > the > > SIP messages originating from MjSip. > > > > FS: http://pastebin.freeswitch.org/8045 > > MjSip: http://pastebin.freeswitch.org/8046 > > > > Thank you very much for your help. > > Best wishes, > > Phil > > > > > > > > > > > On Sun, Mar 29, 2009 at 5:09 PM, wrote: > > > > > > > Hello everyone, > > > > > > > > I am trying to get FS working with the MjSip Java Sip-stack, the > > > SipToSis > > > > source and the normal one. Everything works well within my own > network > > > and > > > > when using x-lite, but when it comes to making calls from MjSip to > an > > > > outside FS server I don't hear any voice - seems to be a NAT problem > or > > > some > > > > kind of other MjSip problem. Registration works fine though and SIP > > > messages > > > > get through ok, but non of the UDP RTP ones. Would be great if > someone > > > could > > > > advice me on how to do the setup correctly. > > > > > > > > The whole FS trace can be found here: > > > http://pastebin.freeswitch.org/8029 > > > > > > > > The settings for MjSip are: > > > > > > > > "via_addr=91.101.58.142 (changed in the whole > trace)","host_port=5090", > > > > "transport_protocols=udp tcp","from_url= > >", > > > > > > > > > > > > > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > > > > > > > > > > > > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > > > > > > > > > > > > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > > > > "bin_rat=rat","bin_vic=vic" > > > > > > > > > > > > Thank you very much. > > > > Best wishes, > > > > Phil > > > > > > > > -- > > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > < > > > MSN%3Aanthony_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 > < > > > sip%3A888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > pstn:213-799-1400 > > > > -- > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > 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:213-799-1400 -- Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01 From msc at freeswitch.org Tue Mar 31 07:13:01 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 31 Mar 2009 07:13:01 -0700 Subject: [Freeswitch-users] Javascript, Hanguphooks, CDRs and User Variables. In-Reply-To: <191c3a030903310559g49211e66x3cbdc1dac088e671@mail.gmail.com> References: <1B99233662E2104983E3550185D3ED734982BE@xena.internal.datapro.co.za> <191c3a030903301512v446d55du5f68a94565851e60@mail.gmail.com> <1B99233662E2104983E3550185D3ED73498352@xena.internal.datapro.co.za> <191c3a030903310559g49211e66x3cbdc1dac088e671@mail.gmail.com> Message-ID: <70D31887-92F2-498C-97D2-AE7797544E97@freeswitch.org> On Mar 31, 2009, at 5:59 AM, Anthony Minessale wrote: > if you set the channel variable 'session_in_hangup_hook=true' early > in the call, the session will be present in your script. > Very cool. I will get this chan var documented on the wiki right away. -MC From anthony.minessale at gmail.com Tue Mar 31 07:42:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 09:42:27 -0500 Subject: [Freeswitch-users] FS - MjSip no voice [SOLVED] SIP 200 / 183 problem In-Reply-To: <20090331140648.227850@gmx.net> References: <20090329220921.227090@gmx.net> <191c3a030903300604h306da82i69a3001ce19aa340@mail.gmail.com> <20090330203353.59630@gmx.net> <191c3a030903301420l78f848ft281430e4d67daa0e@mail.gmail.com> <20090331140648.227850@gmx.net> Message-ID: <191c3a030903310742q15edc6acn8d2f0227e0e2347b@mail.gmail.com> like i said: > maybe that phone does not support early media > > try adding the answer application to your dialplan early media == 183 answer = 200 it depends on your dialplan in FS On Tue, Mar 31, 2009 at 9:06 AM, wrote: > Hello, > > I have found the problem. FS on my local network sends "SIP/2.0 200 OK" > after an invite and FS on the net through the external profil sends > SIP/2.0 183 Session Progress. But MjSip doesn't know how to deal with > 183, so it just ignores the message. For testing I have changed > the 183 header to the 200 one and now it works. > > Thank you for your help and the quick response time. > Best wishes, > Phil > > > >From FS on the net through the external profil: > > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP 90.181.59.141:5090 > ;rport=60315;branch=z9hG4bK256321;received=78.105.17.88 > From: ;tag=z9hG4bK40977269 > To: ;tag=vgg3Zja8pNQcg > Call-ID: 507347917247 at 90.181.59.141 > CSeq: 1 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12839M > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 267 > > v=0 > o=FreeSWITCH 1072777625698755085 8893522831081357051 IN IP4 90.181.59.141 > s=FreeSWITCH > c=IN IP4 91.121.59.148 > t=0 0 > m=audio 26722 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > > > >From FS in my local network: > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.143:5060 > ;rport=5060;branch=z9hG4bK423233;received=192.168.1.102 > From: > >;tag=z9hG4bK42598163 > To: >;tag=Q0X494ZUNaKHH > Call-ID: 961142687222 at 192.168.1.143 > CSeq: 2 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12712M > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 267 > > v=0 > o=FreeSWITCH 5195745633884389954 8941954824002056485 IN IP4 192.168.1.143 > s=FreeSWITCH > c=IN IP4 192.168.1.143 > t=0 0 > m=audio 22680 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > > > > > maybe that phone does not support early media > > > > try adding the answer application to your dialplan > > > > > > On Mon, Mar 30, 2009 at 3:33 PM, wrote: > > > > > Hallo, > > > > > > thank you for your answer Anthony. > > > > > > > > > > > starting at line 192 you seem to be sending yourself a notify, not > > sure > > > > how you did that. > > > > > > That is indeed strange, I have looked at the MjSip code but haven't > > found > > > the cause yet. > > > > > > > you are not by any chance trying to call a registered endpoint using > > the > > > > FS > > > > ip together with @ are you? > > > > say you fs box is 1.2.3.4 and the phone is registered as 1000 > > > > > > > > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4you > > > > would > > > > use sofia/internal/1000%1.2.3.4 > > > > The % tells it to resolve the domain as a locally hosted domain and > > > > translate it to the registered contact instead of using dns. > > > > > > > > > > For testing I at the moment send the incoming call to the voicemail of > > user > > > 1000 with this code: > > > > > > return '''\n'''\ > > > '''\n'''\ > > > '''
\n'''\ > > > '''\n'''\ > > > '''\n'''\ > > > ''' > expression="^(%s)$">\n'''\ > > > '''\n'''\ > > > '''\n'''\ > > > '''\n'''\ > > > '''\n'''\ > > > '''
\n'''\ > > > '''''' % (didNumber, didNumber, id) > > > > > > > > > Works fine with a normal SIP client. > > > I have captured more output with debug enabled and have also captured > > the > > > SIP messages originating from MjSip. > > > > > > FS: http://pastebin.freeswitch.org/8045 > > > MjSip: http://pastebin.freeswitch.org/8046 > > > > > > Thank you very much for your help. > > > Best wishes, > > > Phil > > > > > > > > > > > > > > > On Sun, Mar 29, 2009 at 5:09 PM, wrote: > > > > > > > > > Hello everyone, > > > > > > > > > > I am trying to get FS working with the MjSip Java Sip-stack, the > > > > SipToSis > > > > > source and the normal one. Everything works well within my own > > network > > > > and > > > > > when using x-lite, but when it comes to making calls from MjSip to > > an > > > > > outside FS server I don't hear any voice - seems to be a NAT > problem > > or > > > > some > > > > > kind of other MjSip problem. Registration works fine though and SIP > > > > messages > > > > > get through ok, but non of the UDP RTP ones. Would be great if > > someone > > > > could > > > > > advice me on how to do the setup correctly. > > > > > > > > > > The whole FS trace can be found here: > > > > http://pastebin.freeswitch.org/8029 > > > > > > > > > > The settings for MjSip are: > > > > > > > > > > "via_addr=91.101.58.142 (changed in the whole > > trace)","host_port=5090", > > > > > "transport_protocols=udp tcp","from_url=< > sip:puli at 91.101.58.142:5090 > > > >", > > > > > > > > > > > > > > > > > > > > "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes", > > > > > > > > > > > > > > > > > > > > "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068", > > > > > > > > > > > > > > > > > > > > "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500", > > > > > "bin_rat=rat","bin_vic=vic" > > > > > > > > > > > > > > > Thank you very much. > > > > > Best wishes, > > > > > Phil > > > > > > > > > > -- > > > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > > > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > > > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > > > > > _______________________________________________ > > > > > Freeswitch-users mailing list > > > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > -- > > > > Anthony Minessale II > > > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > > ClueCon http://www.cluecon.com/ > > > > > > > > AIM: anthm > > > > MSN:anthony_minessale at hotmail.com > > > >< > > > > > MSN%3Aanthony_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 > > > >< > > > > > sip%3A888 at conference.freeswitch.org > > > > > > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > > > > > > > > > > > pstn:213-799-1400 > > > > > > -- > > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + > > > Telefonanschluss f?r nur 17,95 Euro/mtl.!* > > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_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 < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > -- > Psssst! Schon vom neuen GMX MultiMessenger geh?rt? Der kann`s mit allen: > http://www.gmx.net/de/go/multimessenger01 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/3cd27146/attachment-0002.html From vince.freeswitch at hightek.org Mon Mar 30 22:12:37 2009 From: vince.freeswitch at hightek.org (Vincent) Date: Tue, 31 Mar 2009 00:12:37 -0500 Subject: [Freeswitch-users] Errors compiling on dragonfly bsd Message-ID: <20090331051237.GA23340@quark.hightek.org> Hi. I'm trying to compile FS on dragonfly BSD 1.10.1-RELEASE. I am compiling the release source from http://files.freeswitch.org/freeswitch-1.0.3.tar.gz I had to write a uname wrapper script to fake being FreeBSD to get configure to complete successfully. I also added "-D__FreeBSD__" to CPPFLAGS to get past an initial compilation error. Also, I initially got the error, "/usr/pkg/bin/bash: line 8: svnversion: command not found" just prior to the line that appears to generate "src/include/switch_swigable_cpp.h". It seems strange that I would need subversion to compile a release source but I installed it anyway in case that error caused other problems. However, now I get the following errors (which I also got prior to installing subversion). Compiling src/switch_apr.c ... src/switch_apr.c: In function `switch_thread_self': src/switch_apr.c:74: warning: implicit declaration of function `apr_os_thread_current' src/switch_apr.c:74: warning: return makes pointer from integer without a cast src/switch_apr.c: In function `switch_thread_rwlock_create': src/switch_apr.c:221: warning: implicit declaration of function `apr_thread_rwlock_create' src/switch_apr.c: In function `switch_thread_rwlock_destroy': src/switch_apr.c:226: warning: implicit declaration of function `apr_thread_rwlock_destroy' ... Pluss a bunch more apr_thread... related warnings Then gmake[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 Making all in src Making all in mod making all mod_amr gmake[5]: *** No rule to make target `/u1/ports/freeswitch-1.0.3/work/freeswitch-1.0.3/libfreeswitch.la', needed by `mod_amr.so'. Stop. gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_amr-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 *** make failed for freeswitch-1.0.3 *** *** Error code 5 Stop in /u1/ports/freeswitch-1.0.3. It does not seem to check the error code either and still gives a "Build Complete" message. Any help would be appreciated. From raffaele.p.guidi at gmail.com Tue Mar 31 00:56:30 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 31 Mar 2009 09:56:30 +0200 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: <20090331072858.GA15463@jdc.jasonjgw.net> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <20090331072858.GA15463@jdc.jasonjgw.net> Message-ID: I am a Yate user and I can tell their mailing list suffer the same problem. My solution? I often ask for help but, as a personal policy, I always write an article or add to an existing one on the project wiki explaining and documenting what people explained to me. This creates a triple value: 1. I have stuff explained 2. other people can find this explanation just googling around 3. I don't have to mantain a separate documentation for myself, I just keep referring to (google and) the project wiki (when I don't exactly remember things I sometimes end reading the wiki and saying "oh, I wrote this!") I suggest, in the end, a kind of "ok I'll give you help but you write this stuff in a piece of documentation" policy. I got the name, too: "The Wiki Tax" ;) Regards, Raffaele On Tue, Mar 31, 2009 at 09:28, Jason White wrote: > Brian West wrote: > > This isn't a buffet where you pull up and demand things be one way or the > > other... this is a community where you start helping. I would love to > > see more helping and less demanding! > > So would I. > > I regularly scan the mailing list looking for questions to answer, but many > of > them relate to scenarios of which I have had no experience, or features > that I > haven't had any reason to use. > > I am worried that the questions are answered by the same people much of the > time, which in the long run will be bad for the project as the community > grows, i.e., more technical support work for the same people (who are > mostly > the core developers as well) is not a sustainable proposition. > > How do other large projects handle this? > > Does anyone have any positive suggestions to offer that would encourage > more > contributions? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/55c588ec/attachment-0002.html From mattdfong at gmail.com Tue Mar 31 08:15:06 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 31 Mar 2009 22:15:06 +0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> Message-ID: <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> Got a few more questions about running LUA scripts, please forgive me, I'm an absolute newbie with LUA. If I want to subscribe to a custom event, and I use con = freeswitch.EventConsumer("CUSTOM my::event"); I get an error. Is this because I must subscribe to the CUSTOM (only) event, and then filter out the events using the Event-Subclass myself? Or am I missing something in the syntax of the subscribe? Also, if I do not have a freeswitch.Session, what is the best way to have my LUA script sleep? I want a functionality, where a statement inside my LUA script gets iterated every 30 seconds. My program does not use a session, so I cannot use session:execute("sleep","1000"), as suggested in the wiki. I tried api::sleep(30000) and a few other combinations with execute but no luck :(. Thanks. --matt On Sat, Mar 28, 2009 at 12:43 AM, Michael Collins wrote: > > con = freeswitch.EventConsumer("all"); > > > > now you have a consumer obj > > > > every time you call con:pop() with no arg you will either get an event or > > nil when there are no events to consume. > > every time you call con:pop(1) the consumer object will block until there > is > > an event. > > > > So you use the first way in conjunction with some other lock to do async > or > > the 2nd way you do a dedicated blocking loop. > > FYI, I added this information to the wiki page for > freeswitch.EventConsumer. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/6dd67954/attachment-0002.html From mike at jerris.com Tue Mar 31 08:32:23 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Mar 2009 11:32:23 -0400 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> Message-ID: as replied earlier, if your doing nothing but consuming events, you can just block instead of sleep: con:pop(1) there is also a msleep function that you can call the same way you do console_log, it takes milli seconds as its arg. Note this should NOT be used when you have a script running as a session, only when you are running an api script. Mike On Mar 31, 2009, at 11:15 AM, Matthew Fong wrote: > Got a few more questions about running LUA scripts, please forgive > me, I'm an absolute newbie with LUA. > > If I want to subscribe to a custom event, and I use > > con = freeswitch.EventConsumer("CUSTOM my::event"); > > I get an error. Is this because I must subscribe to the CUSTOM > (only) event, and then filter out the events using the Event- > Subclass myself? Or am I missing something in the syntax of the > subscribe? > > Also, if I do not have a freeswitch.Session, what is the best way to > have my LUA script sleep? I want a functionality, where a statement > inside my LUA script gets iterated every 30 seconds. My program does > not use a session, so I cannot use session:execute("sleep","1000"), > as suggested in the wiki. I tried api::sleep(30000) and a few other > combinations with execute but no luck :(. > > Thanks. > --matt > > On Sat, Mar 28, 2009 at 12:43 AM, Michael Collins > wrote: > > con = freeswitch.EventConsumer("all"); > > > > now you have a consumer obj > > > > every time you call con:pop() with no arg you will either get an > event or > > nil when there are no events to consume. > > every time you call con:pop(1) the consumer object will block > until there is > > an event. > > > > So you use the first way in conjunction with some other lock to do > async or > > the 2nd way you do a dedicated blocking loop. > > FYI, I added this information to the wiki page for > freeswitch.EventConsumer. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/726287a1/attachment-0002.html From mattdfong at gmail.com Tue Mar 31 08:50:28 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 31 Mar 2009 22:50:28 +0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> Message-ID: <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> I know before I asked about blocking for an event, and maybe I should have created a new topic.. but now I want to actually sleep (rather than block) for a set time frame...this app will not be consuming events. can I get an example of how to use msleep in a lua script? This lua script will be running in the background, and not part of a session or event consumer. Thanks. --matt 2009/3/31 Michael Jerris > as replied earlier, if your doing nothing but consuming events, you can > just block instead of sleep: > con:pop(1) > > there is also a msleep function that you can call the same way you do > console_log, it takes milli seconds as its arg. Note this should NOT be > used when you have a script running as a session, only when you are running > an api script. > > Mike > > On Mar 31, 2009, at 11:15 AM, Matthew Fong wrote: > > Got a few more questions about running LUA scripts, please forgive me, I'm > an absolute newbie with LUA. > If I want to subscribe to a custom event, and I use > > con = freeswitch.EventConsumer("CUSTOM my::event"); > > I get an error. Is this because I must subscribe to the CUSTOM (only) > event, and then filter out the events using the Event-Subclass myself? Or > am I missing something in the syntax of the subscribe? > > Also, if I do not have a freeswitch.Session, what is the best way to have > my LUA script sleep? I want a functionality, where a statement inside my LUA > script gets iterated every 30 seconds. My program does not use a session, so > I cannot use session:execute("sleep","1000"), as suggested in the wiki. I > tried api::sleep(30000) and a few other combinations with execute but no > luck :(. > > Thanks. > --matt > > On Sat, Mar 28, 2009 at 12:43 AM, Michael Collins wrote: > >> > con = freeswitch.EventConsumer("all"); >> > >> > now you have a consumer obj >> > >> > every time you call con:pop() with no arg you will either get an event >> or >> > nil when there are no events to consume. >> > every time you call con:pop(1) the consumer object will block until >> there is >> > an event. >> > >> > So you use the first way in conjunction with some other lock to do async >> or >> > the 2nd way you do a dedicated blocking loop. >> >> FYI, I added this information to the wiki page for >> freeswitch.EventConsumer. >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/beaa744d/attachment-0002.html From mike at jerris.com Tue Mar 31 08:59:08 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Mar 2009 11:59:08 -0400 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> Message-ID: <33A62996-1A30-45F1-95C3-A421867CDFF2@jerris.com> http://wiki.freeswitch.org/wiki/Lua#freeswitch.consoleLog On Mar 31, 2009, at 11:50 AM, Matthew Fong wrote: > I know before I asked about blocking for an event, and maybe I > should have created a new topic.. > > but now I want to actually sleep (rather than block) for a set time > frame...this app will not be consuming events. > > can I get an example of how to use msleep in a lua script? This lua > script will be running in the background, and not part of a session > or event consumer. Thanks. > > --matt > > 2009/3/31 Michael Jerris > as replied earlier, if your doing nothing but consuming events, you > can just block instead of sleep: > > con:pop(1) > > there is also a msleep function that you can call the same way you > do console_log, it takes milli seconds as its arg. Note this should > NOT be used when you have a script running as a session, only when > you are running an api script. > > Mike > > On Mar 31, 2009, at 11:15 AM, Matthew Fong wrote: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/05d2754b/attachment-0002.html From brian at freeswitch.org Tue Mar 31 09:02:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 11:02:26 -0500 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> Message-ID: <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> Lua has no sleep or pause ... if you read thru the lua wiki they show you various ways to accomplish that. On Mar 31, 2009, at 10:50 AM, Matthew Fong wrote: > I know before I asked about blocking for an event, and maybe I > should have created a new topic.. > > but now I want to actually sleep (rather than block) for a set time > frame...this app will not be consuming events. > > can I get an example of how to use msleep in a lua script? This lua > script will be running in the background, and not part of a session > or event consumer. Thanks. > > --ma Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/1de2cd0f/attachment-0002.html From mattdfong at gmail.com Tue Mar 31 09:11:29 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 31 Mar 2009 23:11:29 +0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> Message-ID: <4256bf830903310911o143e2ef5o651415c087c7b3c0@mail.gmail.com> Thanks, the freeswitch.msleep(5000) works! Any comment about the first Q... con = freeswitch.EventConsumer("CUSTOM my::event"); I get an error. Is this because I must subscribe to the CUSTOM (only) event, and then filter out the events using the Event-Subclass myself? Or am I missing something in the syntax of the subscribe? Thanks Michael for your help... --matt 2009/3/31 Brian West > Lua has no sleep or pause ... if you read thru the lua wiki they show you > various ways to accomplish that. > On Mar 31, 2009, at 10:50 AM, Matthew Fong wrote: > > I know before I asked about blocking for an event, and maybe I should have > created a new topic.. > but now I want to actually sleep (rather than block) for a set time > frame...this app will not be consuming events. > > can I get an example of how to use msleep in a lua script? This lua script > will be running in the background, and not part of a session or event > consumer. Thanks. > > --ma > > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/f779b7aa/attachment-0002.html From mike at jerris.com Tue Mar 31 09:18:42 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Mar 2009 12:18:42 -0400 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <4256bf830903310911o143e2ef5o651415c087c7b3c0@mail.gmail.com> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> <4256bf830903310911o143e2ef5o651415c087c7b3c0@mail.gmail.com> Message-ID: http://www.freeswitch.org/docs/class_event_consumer.html#a0 It takes 2 args, not one to specify the subclass Mike On Mar 31, 2009, at 12:11 PM, Matthew Fong wrote: > Thanks, the freeswitch.msleep(5000) works! > > Any comment about the first Q... > > con = freeswitch.EventConsumer("CUSTOM my::event"); > > I get an error. Is this because I must subscribe to the CUSTOM > (only) event, and then filter out the events using the Event- > Subclass myself? Or am I missing something in the syntax of the > subscribe? > > > Thanks Michael for your help... > From msc at freeswitch.org Tue Mar 31 09:24:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Mar 2009 09:24:58 -0700 Subject: [Freeswitch-users] freeswitch.EventConsumer, can be utilized event-based In-Reply-To: <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> References: <4256bf830903270512i2a666086qf59edab0e2c1c094@mail.gmail.com> <191c3a030903270752w25d69562ub391da859818f24e@mail.gmail.com> <87f2f3b90903271043l376a2ae0rec4c113f0ac88516@mail.gmail.com> <4256bf830903310815gc067373y24a71b635f2e343d@mail.gmail.com> <4256bf830903310850j4cc55489k677d4e0c2d3c5db@mail.gmail.com> <30428749-C981-431F-85CE-901881B43B65@freeswitch.org> Message-ID: <87f2f3b90903310924v61f559cfx68809ada6ad8f5ed@mail.gmail.com> FYI, I've documented the msleep method here: http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep I will work on better and more organized API documentation. If anyone out there has time/energy/knowledge of the scripting APIs and is willing to help out please email me off list. -MC 2009/3/31 Brian West > Lua has no sleep or pause ... if you read thru the lua wiki they show you > various ways to accomplish that. > On Mar 31, 2009, at 10:50 AM, Matthew Fong wrote: > > I know before I asked about blocking for an event, and maybe I should have > created a new topic.. > but now I want to actually sleep (rather than block) for a set time > frame...this app will not be consuming events. > > can I get an example of how to use msleep in a lua script? This lua script > will be running in the background, and not part of a session or event > consumer. Thanks. > > --ma > > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/c1bb3cb5/attachment-0002.html From msc at freeswitch.org Tue Mar 31 09:26:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Mar 2009 09:26:35 -0700 Subject: [Freeswitch-users] live iso image with freeswitch In-Reply-To: References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <20090331072858.GA15463@jdc.jasonjgw.net> Message-ID: <87f2f3b90903310926x21b28c7l116fce1a69e8edff@mail.gmail.com> 2009/3/31 Raffaele P. Guidi > I am a Yate user and I can tell their mailing list suffer the same problem. > My solution? I often ask for help but, as a personal policy, I always write > an article or add to an existing one on the project wiki explaining and > documenting what people explained to me. This creates a triple value: > > 1. I have stuff explained > 2. other people can find this explanation just googling around > 3. I don't have to mantain a separate documentation for myself, I > just keep referring to (google and) the project wiki (when I don't exactly > remember things I sometimes end reading the wiki and saying "oh, I wrote > this!") > > I suggest, in the end, a kind of "ok I'll give you help but you write this > stuff in a piece of documentation" policy. I got the name, too: "The Wiki > Tax" ;) > I like it... "The Wiki Tax" -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/f651715e/attachment-0002.html From gkuri at ieee.org Tue Mar 31 09:59:15 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 31 Mar 2009 09:59:15 -0700 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49C45508.402@coppice.org> References: <49C40A83.1050003@ieee.org> <49C45508.402@coppice.org> Message-ID: <49D24BE3.2020101@ieee.org> > This part is interesting, and the subject of a discussion we had > recently. A number of systems try that second re-invite after a 488, but > the SIP specs say the call is pretty much dead after the 488 message is > exchanged. Are they just hoping that maybe the other end will be > non-compliant enough to keep the call alive, and recover its media mode, > or haven't they read the specs? I think they're hoping the other end is willing to recover it's media mode rather than fail the call. I had no idea the call is technically dead after the 488. Honestly, it would be nice if the call would still be recoverable after that 488 on the T.38 ReINVITE, in order to try and negotiate PCMU to try and keep the FAX going, but if that's not how it's supposed to work, I'd rather follow the spec. So for now I've disabled T.38 completely on both the SPA side and I had the carrier disable it on my trunk, so they won't try a T.38 reinvite. Instead they're trying a PCMU ReINVITE and the problem I'm seeing is that if the carrier reINVITEs PCMU after the call initially started out as G729, FS fails the call, because it seems to be trying to transcode between G729 and PCMU, rather than pass the PCMU reINVITE through to the other leg. 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1550 sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/cedarwireless.net/1909XXXXXXX at 65.98.2 36.38 PCMU/8000 20 ms 160 samples 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() Audio params are unchanged for sofia/cedarwireless.net/ 1909XXXXXXX at 65.98.236.38. 2009-03-31 00:30:50 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing Reinvite 2009-03-31 00:30:50 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/cedarwireless.net/1909XXXXXXX at 65.98.236.38 en tering state [completed] 2009-03-31 00:30:50 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/cedarwireless.net/1909XXXXXXX at 65.98.23 6.38 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-03-31 00:30:50 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-31 00:30:50 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! I have inherit_codec=true, set in the dialplan and disable-transcoding=true set in the sofia profile, which is what I thought would do the trick, but it doesn't seem to be doing anything, FS is still trying to transcode between G729 and PCMU. Is there something I'm missing to get the PCMU ReINVITE from one of the legs to passthrough to the other leg? Does this work only in proxy_media or bypass_media modes? I am testing this with the latest rev of trunk as well. Thanks, Gabe From anthony.minessale at gmail.com Tue Mar 31 12:19:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 14:19:18 -0500 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49D24BE3.2020101@ieee.org> References: <49C40A83.1050003@ieee.org> <49C45508.402@coppice.org> <49D24BE3.2020101@ieee.org> Message-ID: <191c3a030903311219h6c825256l7a52720cf9eb3508@mail.gmail.com> correct, we *do not* proxy re-invites except when bypass_media or proxy_media is set. On Tue, Mar 31, 2009 at 11:59 AM, Gabriel Kuri wrote: > > This part is interesting, and the subject of a discussion we had > > recently. A number of systems try that second re-invite after a 488, but > > the SIP specs say the call is pretty much dead after the 488 message is > > exchanged. Are they just hoping that maybe the other end will be > > non-compliant enough to keep the call alive, and recover its media mode, > > or haven't they read the specs? > > I think they're hoping the other end is willing to recover it's media > mode rather than fail the call. I had no idea the call is technically > dead after the 488. Honestly, it would be nice if the call would still > be recoverable after that 488 on the T.38 ReINVITE, in order to try and > negotiate PCMU to try and keep the FAX going, but if that's not how it's > supposed to work, I'd rather follow the spec. > > So for now I've disabled T.38 completely on both the SPA side and I had > the carrier disable it on my trunk, so they won't try a T.38 reinvite. > Instead they're trying a PCMU ReINVITE and the problem I'm seeing is > that if the carrier reINVITEs PCMU after the call initially started out > as G729, FS fails the call, because it seems to be trying to transcode > between G729 and PCMU, rather than pass the PCMU reINVITE through to the > other leg. > > 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1550 > sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU > 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1601 > sofia_glue_tech_set_codec() Set Codec > sofia/cedarwireless.net/1909XXXXXXX at 65.98.2 > 36.38 PCMU/8000 20 ms 160 samples > 2009-03-31 00:30:50 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() > Audio params are unchanged for sofia/cedarwireless.net/ > 1909XXXXXXX at 65.98.236.38. > 2009-03-31 00:30:50 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing Reinvite > 2009-03-31 00:30:50 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/cedarwireless.net/1909XXXXXXX at 65.98.236.38 en > tering state [completed] > 2009-03-31 00:30:50 [DEBUG] switch_core_io.c:655 > switch_core_session_write_frame() > sofia/cedarwireless.net/1909XXXXXXX at 65.98.23 > 6.38 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] > 2009-03-31 00:30:50 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-03-31 00:30:50 [ERR] switch_core_io.c:723 > switch_core_session_write_frame() Codec G.729 decoder error! > > I have inherit_codec=true, set in the dialplan and > disable-transcoding=true set in the sofia profile, which is what I > thought would do the trick, but it doesn't seem to be doing anything, FS > is still trying to transcode between G729 and PCMU. Is there something > I'm missing to get the PCMU ReINVITE from one of the legs to passthrough > to the other leg? Does this work only in proxy_media or bypass_media modes? > > I am testing this with the latest rev of trunk as well. > > Thanks, > > Gabe > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/47736722/attachment-0002.html From gmaruzz at celliax.org Tue Mar 31 13:09:05 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 31 Mar 2009 22:09:05 +0200 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <22800505.post@talk.nabble.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> Message-ID: <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> Ciao Bipin, there is both video and audio. Use vlc (http://www.videolan.org/vlc/) or mplayer (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). 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 2009/3/31 xbipin : > > hi, > > is it just audio or is it that im having broken codecs so cant view any > video? > > Regards, > Bipin > > > > Giovanni Maruzzelli-3 wrote: >> >> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS >> Microsoft Windows >> Download 118MB HD: http://www.celliax.org/final.avi >> >> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p22800505.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Tue Mar 31 13:41:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 15:41:24 -0500 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> Message-ID: <191c3a030903311341r5ff21921l417bd49a095443c@mail.gmail.com> I recommend you reformat this file to be in some format that will play on windows and mac and anything else without having to "get" anything or nobody will watch it =D 2009/3/31 Giovanni Maruzzelli > Ciao Bipin, > there is both video and audio. > > Use vlc (http://www.videolan.org/vlc/) or mplayer > (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). > > 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 > > > > > 2009/3/31 xbipin : > > > > hi, > > > > is it just audio or is it that im having broken codecs so cant view any > > video? > > > > Regards, > > Bipin > > > > > > > > Giovanni Maruzzelli-3 wrote: > >> > >> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS > >> Microsoft Windows > >> Download 118MB HD: http://www.celliax.org/final.avi > >> > >> 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 at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > -- > > View this message in context: > http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p22800505.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/f6bda4ac/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Mar 31 13:44:40 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 31 Mar 2009 21:44:40 +0100 Subject: [Freeswitch-users] Injecting audio into live call Message-ID: Hi Guys, I know this sounds an odd question, but I need to inject audio into an outbound call. The reason for this is that for a pre-paid billing app, I need to let the call initiator know they are running out of credit. Is there a facility to do this? Ideally I only want to let the subscriber, I.e. the one paying for the call to hear this. In other words 'you have 2 minutes left for this call' Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/66901bb6/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Mar 31 13:45:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 31 Mar 2009 21:45:33 +0100 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> Message-ID: Worked for me, just needed to add the missing codec for media player -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: 31 March 2009 21:09 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS and Skypiax on Windows Video How To Ciao Bipin, there is both video and audio. Use vlc (http://www.videolan.org/vlc/) or mplayer (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). 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 2009/3/31 xbipin : > > hi, > > is it just audio or is it that im having broken codecs so cant view any > video? > > Regards, > Bipin > > > > Giovanni Maruzzelli-3 wrote: >> >> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS >> Microsoft Windows >> Download 118MB HD: http://www.celliax.org/final.avi >> >> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p 22800505.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Tue Mar 31 13:51:40 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 15:51:40 -0500 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> Message-ID: <002033F2-5D55-4127-ADBE-8E7A387F7F62@freeswitch.org> You shouldn't have to go get anything :P If you have to spend time to get something to watch the video it sometimes isn't a good thing... have you tried YouTUBE? http://www.perian.org/ /b On Mar 31, 2009, at 3:45 PM, Nik Middleton wrote: > Worked for me, just needed to add the missing codec for media player > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Giovanni Maruzzelli > Sent: 31 March 2009 21:09 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FS and Skypiax on Windows Video How To > > Ciao Bipin, > there is both video and audio. > > Use vlc (http://www.videolan.org/vlc/) or mplayer > (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). > > 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 > > > > > 2009/3/31 xbipin : >> >> hi, >> >> is it just audio or is it that im having broken codecs so cant view > any >> video? >> >> Regards, >> Bipin >> >> >> >> Giovanni Maruzzelli-3 wrote: >>> >>> Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS >>> Microsoft Windows >>> Download 118MB HD: http://www.celliax.org/final.avi >>> >>> 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: > http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p > 22800505.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/36fb2af2/attachment-0002.html From brian at freeswitch.org Tue Mar 31 13:53:00 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 15:53:00 -0500 Subject: [Freeswitch-users] Injecting audio into live call In-Reply-To: References: Message-ID: uuid_displace /b On Mar 31, 2009, at 3:44 PM, Nik Middleton wrote: > Hi Guys, > > I know this sounds an odd question, but I need to inject audio into > an outbound call. The reason for this is that for a pre-paid > billing app, I need to let the call initiator know they are running > out of credit. Is there a facility to do this? Ideally I only want > to let the subscriber, I.e. the one paying for the call to hear > this. In other words ?you have 2 minutes left for this call? > > Regards, > _______ Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/649b3f2d/attachment-0002.html From anthony.minessale at gmail.com Tue Mar 31 14:13:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Mar 2009 16:13:41 -0500 Subject: [Freeswitch-users] Injecting audio into live call In-Reply-To: References: Message-ID: <191c3a030903311413g39786ac5y8ac9106c50dc1235@mail.gmail.com> or, probably better for this situation, uuid_broadcast 2009/3/31 Brian West > uuid_displace > /b > > On Mar 31, 2009, at 3:44 PM, Nik Middleton wrote: > > Hi Guys, > > I know this sounds an odd question, but I need to inject audio into an > outbound call. The reason for this is that for a pre-paid billing app, I > need to let the call initiator know they are running out of credit. Is > there a facility to do this? Ideally I only want to let the subscriber, > I.e. the one paying for the call to hear this. In other words ?you have 2 > minutes left for this call? > > Regards, > _______ > > > Brian West > brian at freeswitch.org > > -- Meet us a ClueCon! http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/e59e58a3/attachment-0002.html From jforman at wcgltd.com Tue Mar 31 15:04:26 2009 From: jforman at wcgltd.com (Josh Forman) Date: Tue, 31 Mar 2009 18:04:26 -0400 Subject: [Freeswitch-users] Greedy codecs on the outbound side Message-ID: I know that on a sofia profile you can set it to be greedy or generous in codec negotiation on the inbound side. How does freeswitch negotate codecs on the outbound side and what ways are available to affect how this takes place? Is there a way to make the codec preferences of the inbound side the codec preference of the outbound side? If not is there a way to set outbound negotations as greedy vs generous? If we disable transcoding in freeswitch would that in effect make the codec negotated on the inbound side be our most prefered codec on the outbound side's codec negotation? Thank you for your help, Josh From badeguruji at yahoo.com Tue Mar 31 16:27:45 2009 From: badeguruji at yahoo.com (badeguruji) Date: Tue, 31 Mar 2009 16:27:45 -0700 (PDT) Subject: [Freeswitch-users] using freeswitch on ubuntu in home network Message-ID: <902786.43554.qm@web80408.mail.mud.yahoo.com> Hello, Friends, can someone please help me understand what hardware and/or software (like FreeSwitch) do i need, to have a working setup like below: DSL from telco----->switch+NAT------>ubuntu server |__________>voip phone |___________>normal PSTN phone Q. what voip related software will i need to run above setup? Q. what hardware (like phone card etc.) do i need in my PC? Q. will i be able to use my existing phone number? (i do not have voip service right now) Q. will i be able to use my regular phone? or do i need voip phone? I am looking for these featuers: 1. full call records. 2. visual voicemail 3. seperate voicemail box for each family member. 4. call forwarding on each user basis. 5. ability to make more then one outbound call, with having ONLY one home phone number. thank you, Rajeev ________________________________ ~~aapka kalyan ho~~ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/0a45a198/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 19861 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/0a45a198/attachment-0002.jpe From jason at jasonjgw.net Tue Mar 31 16:36:43 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Apr 2009 10:36:43 +1100 Subject: [Freeswitch-users] contributing to the wiki In-Reply-To: <87f2f3b90903310926x21b28c7l116fce1a69e8edff@mail.gmail.com> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <20090331072858.GA15463@jdc.jasonjgw.net> <87f2f3b90903310926x21b28c7l116fce1a69e8edff@mail.gmail.com> Message-ID: <20090331233643.GA7764@jdc.jasonjgw.net> Michael Collins wrote: > I like it... "The Wiki Tax" It's an excellent suggestion. As an aside, would it be possible for the wiki administrator to modify the configuration so that there is a means of subscribing without having to deal with a captcha? For people who can't see the screen, this is a real barrier to participation. From sprice at gmail.com Tue Mar 31 16:47:53 2009 From: sprice at gmail.com (SP) Date: Tue, 31 Mar 2009 18:47:53 -0500 Subject: [Freeswitch-users] Greedy codecs on the outbound side In-Reply-To: References: Message-ID: <7e2ac3270903311647g28dc9b4ck10a8ab7cd47aa89f@mail.gmail.com> http://wiki.freeswitch.org/wiki/Codec_negotiation http://lmgtfy.com/?q=freeswitch+late+negotiation On Tue, Mar 31, 2009 at 17:04, Josh Forman wrote: > I know that on a sofia profile you can set it to be greedy or generous > in codec negotiation on the inbound side. > > How does freeswitch negotate codecs on the outbound side and what ways > are available to affect how this takes place? > Is there a way to make the codec preferences of the inbound side the > codec preference of the outbound side? > If not is there a way to set outbound negotations as greedy vs generous? > If we disable transcoding in freeswitch would that in effect make the > codec negotated on the inbound side be our most prefered codec on the > outbound side's codec negotation? > > Thank you for your help, > > Josh > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From brian at freeswitch.org Tue Mar 31 16:49:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 18:49:06 -0500 Subject: [Freeswitch-users] contributing to the wiki In-Reply-To: <20090331233643.GA7764@jdc.jasonjgw.net> References: <22784622.post@talk.nabble.com> <225BE171-7357-4E5C-97CB-BFCBDFF34440@freeswitch.org> <20090331072858.GA15463@jdc.jasonjgw.net> <87f2f3b90903310926x21b28c7l116fce1a69e8edff@mail.gmail.com> <20090331233643.GA7764@jdc.jasonjgw.net> Message-ID: <76A67C51-ACDB-4916-9ADE-054D53647AA2@freeswitch.org> On Mar 31, 2009, at 6:36 PM, Jason White wrote: > It's an excellent suggestion. > > As an aside, would it be possible for the wiki administrator to > modify the > configuration so that there is a means of subscribing without having > to deal > with a captcha? Not sure we can do that but I suspect we could flag it in such a way where some users can edit without it... I totally understand the need! ;) > > For people who can't see the screen, this is a real barrier to > participatio Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/dc573141/attachment-0002.html From jason at jasonjgw.net Tue Mar 31 16:49:39 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Apr 2009 10:49:39 +1100 Subject: [Freeswitch-users] using freeswitch on ubuntu in home network In-Reply-To: <902786.43554.qm@web80408.mail.mud.yahoo.com> References: <902786.43554.qm@web80408.mail.mud.yahoo.com> Message-ID: <20090331234939.GA7922@jdc.jasonjgw.net> badeguruji wrote: > Q. what voip related software will i need to run above setup? FreeSWITCH > Q. what hardware (like phone card etc.) do i need in my PC? A digium or similar card, and a SIP phone of course. > Q. will i be able to use my existing phone number? (i do not have voip service right now) > Q. will i be able to use my regular phone? or do i need voip phone? That depends on your SIP provider and the laws in your country regarding transferrability of phone numbers. > > I am looking for these featuers: > 1. full call records. > 2. visual voicemail > 3. seperate voicemail box for each family member. > 4. call forwarding on each user basis. > 5. ability to make more then one outbound call, with having ONLY one home phone number. That shouldn't be difficult. From brian at freeswitch.org Tue Mar 31 16:52:48 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Mar 2009 18:52:48 -0500 Subject: [Freeswitch-users] using freeswitch on ubuntu in home network In-Reply-To: <902786.43554.qm@web80408.mail.mud.yahoo.com> References: <902786.43554.qm@web80408.mail.mud.yahoo.com> Message-ID: First off please no HTML emails.... We have a large community of blind users that might have a hard time reading your messages or replying which they do all the time! ;) Thanks guys. On Mar 31, 2009, at 6:27 PM, badeguruji wrote: > Hello, > > Friends, can someone please help me understand what hardware and/or > software (like FreeSwitch) do i need, to have a working setup like > below: > > DSL from telco----->switch+NAT------>ubuntu server > |__________>voip phone > |___________>normal PSTN phone > > Q. what voip related software will i need to run above setup? You'll need FreeSWITCH, Voip Phone and a PSTN interface. > Q. what hardware (like phone card etc.) do i need in my PC? Various interface cards are outlined on the wiki... http://wiki.freeswitch.org > Q. will i be able to use my existing phone number? (i do not have > voip service right now) If you get a card to interface your server to the PSTN then yes you'll be able to use it. > Q. will i be able to use my regular phone? or do i need voip phone? If it can plugin to an ATA or other such device to let you ring a standard phone. > > I am looking for these featuers: > 1. full call records. Yes > 2. visual voicemail Yes. Via web. > 3. seperate voicemail box for each family member. Yes > 4. call forwarding on each user basis. Yes > 5. ability to make more then one outbound call, with having ONLY one > home phone number. You need a voip provider or one line per concurrent call you wish to make. > > > thank you, > Rajeev > ________________________________ > ~~aapka kalyan ho~~ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Brian West brian at freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com From msc at freeswitch.org Tue Mar 31 17:10:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Mar 2009 17:10:58 -0700 Subject: [Freeswitch-users] Call For Help: Janitor Projects Message-ID: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a "janitor projects" wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say "janitor" projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of "janitors" and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090331/a0ee2cb7/attachment-0002.html From mitul at enterux.com Tue Mar 31 18:36:36 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 1 Apr 2009 07:06:36 +0530 Subject: [Freeswitch-users] using freeswitch on ubuntu in home network In-Reply-To: <20090331234939.GA7922@jdc.jasonjgw.net> References: <902786.43554.qm@web80408.mail.mud.yahoo.com> <20090331234939.GA7922@jdc.jasonjgw.net> Message-ID: <7A8719B0-F715-45B4-885F-E3ABBE94701E@enterux.com> Hello, You can't make more than one outbound call if you have only one phone line from bsnl. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 01-Apr-09, at 5:19, Jason White wrote: > badeguruji wrote: > >> Q. what voip related software will i need to run above setup? > > FreeSWITCH >> Q. what hardware (like phone card etc.) do i need in my PC? > > A digium or similar card, and a SIP phone of course. >> Q. will i be able to use my existing phone number? (i do not have >> voip service right now) >> Q. will i be able to use my regular phone? or do i need voip phone? > > That depends on your SIP provider and the laws in your country > regarding > transferrability of phone numbers. >> >> I am looking for these featuers: >> 1. full call records. >> 2. visual voicemail >> 3. seperate voicemail box for each family member. >> 4. call forwarding on each user basis. >> 5. ability to make more then one outbound call, with having ONLY >> one home phone number. > > That shouldn't be difficult. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Tue Mar 31 22:50:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 1 Apr 2009 07:50:36 +0200 Subject: [Freeswitch-users] FS and Skypiax on Windows Video How To In-Reply-To: <002033F2-5D55-4127-ADBE-8E7A387F7F62@freeswitch.org> References: <7b197bef0903310026g51eebb28hcd7035b2e26eda63@mail.gmail.com> <22800505.post@talk.nabble.com> <7b197bef0903311309k52208d99pd650cfb75e2a4c44@mail.gmail.com> <002033F2-5D55-4127-ADBE-8E7A387F7F62@freeswitch.org> Message-ID: <7b197bef0903312250s2afe6346r8da13c848a98e3b9@mail.gmail.com> Oooops, I was not aware you cannot see the video on Windows (I use mplayer and vlc on windows, and never bother to start windows media player :-) ). I agree that the best would be youtube or so. I don't know how to upload video on youtube, and I'll be not in my office for a week. Can one of you kind souls upload the video to youtube? It would be soooo nice! I'll try to do that when I'm back if nobody steps out. gm 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 2009/3/31 Brian West : > You shouldn't have to go get anything :P ?If you have to spend time to get > something to watch the video it sometimes isn't a good thing... have you > tried YouTUBE? > http://www.perian.org/ > /b > On Mar 31, 2009, at 3:45 PM, Nik Middleton wrote: > > Worked for me, just needed to add the missing codec for media player > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Giovanni Maruzzelli > Sent: 31 March 2009 21:09 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FS and Skypiax on Windows Video How To > > Ciao Bipin, > there is both video and audio. > > Use vlc (http://www.videolan.org/vlc/) or mplayer > (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-). > > 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 > > > > > 2009/3/31 xbipin : > > hi, > > is it just audio or is it that im having broken codecs so cant view > > any > > video? > > Regards, > > Bipin > > > > Giovanni Maruzzelli-3 wrote: > > Kulwinder Singh contributed this HOW TO: Freeswitch & Skype- OS > > Microsoft Windows > > Download 118MB HD: http://www.celliax.org/final.avi > > 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > > View this message in context: > > http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p > 22800505.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Brian West > brian at freeswitch.org > -- Meet us a ClueCon! ?http://www.cluecon.com > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mszlazak at aol.com Tue Mar 31 23:21:54 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 01 Apr 2009 02:21:54 -0400 Subject: [Freeswitch-users] Call For Help: Janitor Projects In-Reply-To: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> References: <87f2f3b90903311710p67dc5433x135bebef21f8b9b7@mail.gmail.com> Message-ID: <8CB80A9498DAAC9-7D8-2FB@webmail-mf17.sysops.aol.com> First off. I would not call it a "janitors project" since that may offend some. A second problem is your notion that documentation is "not-quite-as-important" a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of "customer service." Good customer service is then a part of "sales and marketing." Much more often than not, It's sales and marketing that is more important to making something a "real product"? than engineering. "Build it and they will come" almost never works. Anyway, I think you need a new name for this project. -----Original Message----- From: Michael Collins To: freeswitch-users at lists.freeswitch.org ; freeswitch-dev at lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a "janitor projects" wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say "janitor" projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of "janitors" and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009!? http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090401/2c867d94/attachment-0002.html